How do you configure Cisco Call Manager 4.0 to work with an E&M Trunk on a 2600 series with a NM-2V and E&M Module?

I have no problem programming the router, the problem is adding the router as a gateway from cisco call manager. Although configuration for programming the router probably wouldn't hurt. Currently here is my config, I read through the manuals and I know you need to use type 2 E&M Signaling.

interface Ethernet0/0
 ip address 10.0.0.32 255.255.255.0
 full-duplex
!
interface Serial0/0
 no ip address
 shutdown
 no fair-queue
!
ip classless
ip http server
!
!
!
voice-port 1/0/0
 cptone CA
 operation 4-wire
 type 2
!
voice-port 1/0/1
 cptone CA
 operation 4-wire
 type 2
!
dial-peer cor custom
!

Also here is a link in reagrds to your question:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00801ecfc2.html#49714

Is this your first Cisco Call manager setup? Is this for a client or inhouse lab?

What are you hooking up in regards to the E&M circuit? E&M from a provider or another PBX?

This is for our office lab. I have gone through most of the documentation for call manager already, if you look on that link, you will see that there is nothing on adding an E&M gateway, that is my biggest problem right now. The only choice I have is FXO/FXS ports on an NV-2M

I am hooking up a Norstar MICS to the E&M of a cisco router. Here is also how I have my wiring configuration:

http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a00800942ef.shtml

I have the 25 wire Amphenol Cable wired up like this FYI. I also plugged this into the 2ICS/AUX slot. My E&M/DISA card is in Slot 2 of my MICS.

I have 2 Ports that run to the VIC 2E/M card with the following wire configurations:

I followed the MICS guide on wiring these:

RJPIN  Service Wire-Color
PORT1
1 SB Brown-White
2 M  White-Brown
3
4
5
6
7 E White Green
8 SG Green With

Port 2
1 SB Green Red
2 M Red Green
3
4
5
6
7 E Red Orange
8 SG Orange Red

I am guessing that Cisco Call Manager 4.0 is not compatible with E&M on a 2600 series? Maybe I have to buy the T1 Trunks for both the Norstar MICS and the Cisco Router? Or maybe I just have to buy some other equipment who knows?

Patience first, second let look at what you are doing:

Providing E&M TIE Trunks between a Cisco Call Maanger and a Nortel MICS. Is this correct?

Yes that is exactly what I am trying to do.

Also what versionof MICS software do you have?

System version 30mdF01 NAT I just bought this about 1 month ago, but I am not sure where to look for the version number.

Would you mind a few pointers that will help you out?

Please! Not at all you have been more then helpful.

I would love some pointers i mean! I worded that funny

Okay, then:

Need to break things down to the following groups:

Components- equipmnet type and software versions
Connectivity- make sure cabling is correct
Signaling- make sure that PBX/Cisco using same signaling
Programming- program each system to access TIE lines

seems simple but most folks forget to do the simple things sometimes to show that they have so much knowledge. I will digress for a second with a small story. In the 80's I was a refrigeration tech, buddy bought a resturaunt, to help him out I rebult a few ice machines for him. Well finish rebuilding the last one, turned the switch on, no go? Spent two hours ripping everything out to have someone come and tell me that I forgot to plug the darn thing in! Okay enough said.

Do you have any docs for the MICS?

I have all the docs for the MICS, the probablem right now is Call Manager though, because I can't even setup the E&M settings.

What I am trying to do now is to program the Cisco Call Manager 4.0 to access the E&M Tie line on a cisco 2610 Router with an E&M Card. After that is done the rest shouldn't be that difficult I think.

I know that you want to get the Call Manager going, but to be blunt even if you did, my money is on that it will not work. Your link above for the wiring is for an Meridain Option 11, total different beast than the Norstar. Also, different versions of the Norstar only allowwed certian cartridges in certian slots, if you need to move the card to another module the wiring would change again.

So you can see why I follow my simple philosphy above. Also in regards to the Cisco Call Manager it does support E&M, just its almost 2 in the morning and just cannot remeber the exact config (I did notice that you had the E&M set for type 2, which might not work with the MICS).

So you need to enter programing of the MICS to obtain the version number.

Yanked out the card it says v 5.01

Just remembered, Call manger supports E&M but the NV-2M does not:

http://www.cisco.com/en/US/products/hw/routers/ps259/products_tech_note09186a00800e73f6.shtml

That is one way to do it, 5.0 allows to put the E&M in slots 1,2,3 so no problems in slot 2

Actually if you look under 261x it is supported. VIC-2E&M is supported by the NM-2V but not the NM-HDV. I can even configure it in the router. Go figure...

So my wiring connections are in correct? I followed the manual in the back of the book.

Just one question though, here is the footnot from link: Look at line 4 (just making sure)

4The NM-HD-1V and NM-HD-2V have feature parity with the NM-1V and NM-2V. However, there is currently no support on NM-1V and NM-2V for:

Trunked ATM Adaptation Layer 2 (AAL2) on analog ports

Universal Foreign Exchange Office (FXO) (VIC2-xFXO) voice cards with software selectable CAMA

Any VIC2 cards, such as VIC2-2BRI-NT/TE, VIC2-2FXS, VIC2-2E/M, and VIC-4FXS/DID cards

The drop of DSPs on a POTS-to-POTS call that stays within the network module

Currently no support

"Any VIC2 cards, such as VIC2-2BRI-NT/TE, VIC2-2FXS, VIC2-2E/M, and VIC-4FXS/Direct Inward Dial (DID) cards" well it looks like they may not be compatible hmmm do you think this is the case? It is quite strange that I can acutally configure the card in the router...

Enough for one night time to go home! This thread is getting longer and longer...

Still there?

Just going thru the post and realized that you did not list a Trunk Module, but have the E&M in slot 2 of the MICS? This is not supported, you plug the E&M into any slot of the TM but not the MICS, the main reason is for the wiring layout.

Go to the Instllers guide of the MICS book, Look for section called Installing the Cartridges, the table in that section shows what can be plugged into what slot.

Heading to bed, catch up with you in a few hours.

Harry,

All set, lets go!

I am back! I am just reading this carefully, amazing what a night of sleep allows

4The NM-HD-1V and NM-HD-2V have feature parity with the NM-1V and NM-2V. However, there is currently no support on NM-1V and NM-2V for:

Trunked ATM Adaptation Layer 2 (AAL2) on analog ports

Universal Foreign Exchange Office (FXO) (VIC2-xFXO) voice cards with software selectable CAMA

Any VIC2 cards, such as VIC2-2BRI-NT/TE, VIC2-2FXS, VIC2-2E/M, and VIC-4FXS/Direct Inward Dial (DID) cards

The drop of DSPs on a Plain Old Telephone Service (POTS)-to-POTS call that stays within the network module.


I model of card I have is a VIC-2E/M not a VIC2-2E/M which would explain why it works. Now I just have to figureout how to get call manager to recongize it.

Just for the record what model 2600 and IOS version are you using?

Also might have a bead on the issue, I have not seen the use of MGCP with E&M, but have seen the use of H.323 mode with E&M. Trying to confirm this, but more than likely need to configure 2600 as a H.323 gateway.

Plus could you post your router config in its entirety?

Easy does it boys ....  Remember the basics here:

1. The only protocol with which Call Manager has control of a gateway is MGCP.
2. H.323 is unaware of Call Manager, and Call Manager is not required to make an H.323 gateaway function properly.

Therefore, we should first verify call setup and teardown between the router and the MICS and then verify registration of the gateway with Call Manager.  All that is required is proper ip addressing to accomplish this.

So the question rwotherspoon is that will the 2600 with above equipment using MGCP be able to reconize the E&M ports?

I looked at this thread in google

http://groups.google.com/groups?q=norstar+e%26m+mics+seize+line&hl=en&lr=&selm=478c6a3a.0112041001.2ee503da%40posting.google.com&rnum=1

Went through the debugging steps
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f5e.shtml#step2

How do I configure the MICS to seize the line? I am reading the manual, maybe I am missing something.

Here is my 2600 Config

Cisco Internetwork Operating System Software
IOS (tm) C2600 Software (C2600-JK8O3S-M), Version 12.2(12), RELEASE SOFTWARE (fc1)
Copyright (c) 1986-2002 by cisco Systems, Inc.
Compiled Wed 21-Aug-02 02:10 by pwade
Image text-base: 0x8000808C, data-base: 0x815AC1C0

ROM: System Bootstrap, Version 11.3(2)XA4, RELEASE SOFTWARE (fc1)

cscscovoipgw01 uptime is 3 days, 8 hours, 11 minutes
System returned to ROM by power-on
System image file is "flash:/c2600-jk8o3s-mz.122-12.bin"

cisco 2610 (MPC860) processor (revision 0x203) with 61440K/4096K bytes of memory.
Processor board ID JAD034502S1 (2656574567)
M860 processor: part number 0, mask 49
Bridging software.
X.25 software, Version 3.0.0.
SuperLAT software (copyright 1990 by Meridian Technology Corp).
TN3270 Emulation software.
1 Ethernet/IEEE 802.3 interface(s)
1 Serial network interface(s)
2 Voice E & M interface(s)
32K bytes of non-volatile configuration memory.
16384K bytes of processor board System flash (Read/Write)

Configuration register is 0x2102


version 12.2
service timestamps debug uptime
service timestamps log uptime
service password-encryption
!
hostname cscscovoipgw01
!
enable secret 5 $1$L1Om$uZtzG2jtYAPtq1s4PQ0Q7/
!
ip subnet-zero
!
!
!
ip audit notify log
ip audit po max-events 100
!
call rsvp-sync
!
!
!
!
!
!
!
!
interface Ethernet0/0
 ip address 10.0.0.32 255.255.255.0
 full-duplex
!
interface Serial0/0
 no ip address
 shutdown
 no fair-queue
!
ip classless
ip http server
!
!
!
voice-port 1/0/0
 cptone CA
 operation 4-wire
 type 2
!
voice-port 1/0/1
 cptone CA
 operation 4-wire
 type 2
!
dial-peer cor custom
!
!
!
!
banner motd ^CComet Cisco 2610 VOIP Gateway^C
!
line con 0
 password 7 072E057A195B4E5344
 login
line aux 0
 password 7 013222320C59515972
 login
line vty 0 4
 password 7 080068785E4B524141
 login
!
end

Harry,

Sorry for the delay, I have been hit by a really bad computer virus, working to resolve it. But just off hand on the MICS, you should program the E&M lines into a pool/trunk group. Then you assign a number that allows you to access that pool/trunk group, lets say 8 (common to use for TIE trunks). To verify that the PBX is programmed correctly you can wire the E&M ports together on the MICS (wire port 1 directly to port 2). This will loop the call right back into the MICS, if this is not working then the something is wrong on the PBX side.

I will update more later today, once i resolve this virus, nasty little bugger.

Got the config working and i am able to seize the line from the MICS pressing 8. I assigned the E&M lines to Pool B and assigned them to the number 8. I guess this only thing left now is to configure the cisco 2610 Router to be a H.323 Gateway between the CallManager System and the MICS system. This should be interesting.

I have sucessfully setup the 2600 series as an H.323 Gateway and created the proper routing in cisco call manager. I created a h.323 gateway and also added a route pattern/ Hunt Group. There are only 2 issues left that I haven't figured out:

1) On the Norstar, I setup the E&M Lines so that you have to press 8 before it dials out. Is there any way I can setup the MICS so that when I press the extension (200 on the ip phone) 200 instead of pressing 8 first and then pressing 200? I don't want to have to deal with the dial tone.

2) I am unable to call back from the call manager to the norstar MICS. I have it setup right now so that the MICS E&M line is auto answer with DISA disabled. I have also tried it with manual. The following is a config from the 2600 gateway. Almost there! Next step is T1!

SH RUN

version 12.2
service timestamps debug uptime
service timestamps log uptime
service password-encryption
!
hostname cscscovoipgw01
!
enable secret 5 $1$L1Om$uZtzG2jtYAPtq1s4PQ0Q7/
!
ip subnet-zero
!
!
!
ip audit notify log
ip audit po max-events 100
!
call rsvp-sync
!
!
!
voice class h323 1
 h225 timeout tcp establish 3
!
!
!
!
!
!
interface Ethernet0/0
 ip address 10.100.0.32 255.255.255.0
 full-duplex
 h323-gateway voip interface
 h323-gateway voip h323-id cscovoipgw01@cometcomputing.com
!
interface Serial0/0
 no ip address
 shutdown
 no fair-queue
!
ip classless
ip http server
!
!
!
voice-port 1/0/0
 cptone CA
 operation 4-wire
 type 2
!
voice-port 1/0/1
 cptone CA
 operation 4-wire
 type 2
!
dial-peer cor custom
!
!
!
dial-peer voice 100 voip
 preference 1
 destination-pattern 20.
 voice-class h323 1
 session target ipv4:10.100.0.40
 dtmf-relay h245-alphanumeric
 ip precedence 5
!
dial-peer voice 1 pots
 destination-pattern 22.
 no digit-strip
 port 1/0/0
!
dial-peer voice 2 pots
 destination-pattern 22.
 no digit-strip
 port 1/0/1
!
gateway
!
!
banner motd ^CComet Cisco 2610 VOIP Gateway^C
!        
line con 0
 password 7 072E057A195B4E5344
 login
line aux 0
 password 7 013222320C59515972
 login
line vty 0 4
 password 7 080068785E4B524141
 login
!
end


SH VOICE PORT 1/0/0

cscscovoipgw01#sh voice port 1/0/0

recEive And transMit 1/0/0 Slot is 1, Sub-unit is 0, Port is 0
 Type of VoicePort is E&M
 Operation State is DORMANT
 Administrative State is UP
 No Interface Down Failure
 Description is not set
 Noise Regeneration is enabled
 Non Linear Processing is enabled
 Music On Hold Threshold is Set to -38 dBm
 In Gain is Set to 0 dB
 Out Attenuation is Set to 0 dB
 Echo Cancellation is enabled
 Echo Cancellation NLP mute is disabled
 Echo Cancellation NLP threshold is -21 dB
 Echo Cancel Coverage is set to 8 ms
 Playout-delay Mode is set to default
 Playout-delay Nominal is set to 60 ms
 Playout-delay Maximum is set to 200 ms
 Playout-delay Minimum mode is set to default, value 40 ms
 Connection Mode is normal
 Connection Number is not set
 Initial Time Out is set to 10 s
 Interdigit Time Out is set to 10 s
 Ringing Time Out is set to 180 s
 Companding Type is u-law
 Region Tone is set for CA

 Analog Info Follows:
 Currently processing none
 Maintenance Mode Set to None (not in mtc mode)
 Number of signaling protocol errors are 0
 Impedance is set to 600r Ohm
 Wait Release Time Out is 30 s
 Station name None, Station number None

 Voice card specific Info Follows:
 Signal Type is wink-start
 Operation Type is 4-wire
 E&M Type is 2
 Dial Type is dtmf
 In Seizure is inactive
 Out Seizure is inactive
 Digit Duration Timing is set to 100 ms
 InterDigit Duration Timing is set to 100 ms
 Pulse Rate Timing is set to 10 pulses/second
 InterDigit Pulse Duration Timing is set to 750 ms
 Clear Wait Duration Timing is set to 400 ms
 Wink Wait Duration Timing is set to 200 ms
 Wait Wink Duration Timing is set to 550 ms
 Wink Duration Timing is set to 200 ms
 Delay Start Timing is set to 300 ms
 Delay Duration Timing is set to 2000 ms
 Dial Pulse Min. Delay is set to 140 ms
 Percent Break of Pulse is 60 percent
 Auto Cut-through is disabled
 Dialout Delay is 70 ms


cscscovoipgw01#sh dialplan number 222
Dial string terminator: #
Macro Exp.: 222

VoiceEncapPeer2
        information type = voice,
        tag = 2, destination-pattern = `22.',
        answer-address = `', preference=0,
        numbering Type = `unknown'
        group = 2, Admin state is up, Operation state is up,
        incoming called-number = `', connections/maximum = 0/unlimited,
        DTMF Relay = disabled,
        huntstop = disabled,
        in bound application associated: DEFAULT
        out bound application associated:
        permission :both
        incoming COR list:maximum capability
        outgoing COR list:minimum requirement
        type = pots, prefix = `',
        forward-digits default
        session-target = `', voice-port = `1/0/1',
        direct-inward-dial = disabled,
        digit_strip = disabled,

        register E.164 number with GK = TRUE
        Connect Time = 0, Charged Units = 0,
        Successful Calls = 0, Failed Calls = 8, Incomplete Calls = 0
        Accepted Calls = 0, Refused Calls = 1,
        Last Disconnect Cause is "1C  ",
        Last Disconnect Text is "invalid number.",
        Last Setup Time = 165714.
Matched: 222   Digits: 2
Target:

VoiceEncapPeer1
        information type = voice,
        tag = 1, destination-pattern = `22.',
        answer-address = `', preference=0,
        numbering Type = `unknown'
        group = 1, Admin state is up, Operation state is up,
        incoming called-number = `', connections/maximum = 0/unlimited,
        DTMF Relay = disabled,
        huntstop = disabled,
        in bound application associated: DEFAULT
        out bound application associated:
        permission :both
        incoming COR list:maximum capability
        outgoing COR list:minimum requirement
        type = pots, prefix = `',
        forward-digits default
        session-target = `', voice-port = `1/0/0',
        direct-inward-dial = disabled,
        digit_strip = disabled,

        register E.164 number with GK = TRUE
        Connect Time = 7029, Charged Units = 0,
        Successful Calls = 15, Failed Calls = 0, Incomplete Calls = 0
        Accepted Calls = 5, Refused Calls = 3,
        Last Disconnect Cause is "10  ",
        Last Disconnect Text is "normal call clearing.",
        Last Setup Time = 166781.
Matched: 222   Digits: 2
Target:

cscscovoipgw01#sh dialplan number 200
Dial string terminator: #
Macro Exp.: 200

VoiceOverIpPeer100
        information type = voice,
        tag = 100, destination-pattern = `20.',
        answer-address = `', preference=1,
        numbering Type = `unknown'
        group = 100, Admin state is up, Operation state is up,
        incoming called-number = `', connections/maximum = 0/unlimited,
        DTMF Relay = enabled,
        modem passthrough = system,
        huntstop = disabled,
        in bound application associated: DEFAULT
        out bound application associated:
        permission :both
        incoming COR list:maximum capability
        outgoing COR list:minimum requirement
        type = voip, session-target = `ipv4:10.100.0.40',
        technology prefix:
        settle-call = disabled
        ip precedence = 5, UDP checksum = disabled,
        session-protocol = cisco, session-transport = udp, req-qos = best-effort,
        acc-qos = best-effort,
        dtmf-relay = h245-alphanumeric,
        fax rate = voice,   payload size =  20 bytes
        fax protocol = system
        fax-relay ecm enable
        fax NSF = 0xAD0051 (default)
        codec = g729r8,   payload size =  20 bytes,
        Expect factor = 0, Icpif = 20,
        Playout Mode is set to default,
        Initial 60 ms, Max 200 ms
        Playout-delay Minimum mode is set to default, value 40 ms
        Max Redirects = 1, signaling-type = cas,
        CLID Restrict = disabled
        VAD = enabled, Poor QOV Trap = disabled,
        voice class perm tag = `'
        Connect Time = 5845, Charged Units = 0,
        Successful Calls = 4, Failed Calls = 0, Incomplete Calls = 0
        Accepted Calls = 15, Refused Calls = 0,
        Last Disconnect Cause is "10  ",
        Last Disconnect Text is "normal call clearing.",
        Last Setup Time = 166921.
Matched: 200   Digits: 2
Target: ipv4:10.100.0.40

Harry,

Hope to jump on this in the morning or Monday. In reagrds to question #1, are you looking ring extenstion 200 which happens to be one of the Cisco IP phones coonected to the Cisco call Manager without dialing 8 first? The short answer I belive is yes, there would be a few caviets: 200 could not exsist on the MICS and what ever else i figure out in the morning, still battling the worms and viruses.

More to come later. Glad to hear that things are moving forward.

Currently right now I have also installed an FXS card in the 2600 series router to help debug why I am not able to call the Nortel PBX System. Currently, I have no problem calling from

1) Call Manager to the FXS Phone
2) FXS Phone to the Call Manager
3) Nortel MICS to the FXS Phone
4) Nortel MICE to the Call Manager (pressing 8 on line pool B, switching back to the public pool I can't get it to work)

So right now the only problem I have is not being able to send call the Nortel MICS extensions from the FXS port or the Call Manager. The only line that picks up is the prime line when I set the AutoAnswer to yes or manual, the extensions aren't being routed. Any Ideas?

Can you clarify the last post?

Basically, when i place a call to the norstar system, the only line that picks up is the prime line. For example, on my norstar i have an extension 222. When I call 222, it always routes to the prime line 221 instead of going to 222.

So from the Cisco phone you dial 222, which is routing thru Cisco Router on the FXS port correct? Question is then, the FXS port from the Cisco to the MICS is that terminated into the system as an exntestion or CO port? If the Cisco is terminated into MICS on the CO port then, the system is doing what it is supposed to do by deafult. CO Line 1 is deafulted to ring 221.

What you need to do if the case above is correct, would be to terminate FXS port from Cisco router onto MICS extenstion, then Cisco port would need to be configured to go of hook when it recieved digits 222 from Cisco phone. When the port goes off hook its drwing local dial tone from the MICS, then you would pulse the digits 222.

I am using E&M right now.

All my cisco gateway does when it receives the digits 22x is that if fowards the out of E&M Voice Port 1/0/0 and 1/0/1. So I am missing something in my configuration on my router? When I call 20x from the FXS port to my cisco call manager, it works perfect, the only problem I have is dialing into the E&M trunk on the MICS.

Does E&M Not allow you to dial an extension after you seize the line from the cisco router?

No should be able too. How do you have the E&M set up? Immeadite or Wink?

I have tried both immediate and wink-start, they both work but it always forwards to the prime line and not dialing in directly to the extension.

Have you tried putting a pause in front of the dial string once you sieze the line?

The MICS might not be recieveing the full digit string due to some timing issues.

I think it may be a timing issue as well, maybe it is sending too fast, I will put a pause to see what happens.

Use wink start with a pause.

You mean configuration of this on the norstar or the router?

Just a quick question on the norstar MICS, i read that each E&M card has 2 lines that are for the trunks and 2 that are DTMF receivers. I don't have the DTMF receivers hooked up to anything because there is no documentation on this, do I need to hook the DTMF receivers to anything? What are those 2 extra lines used for?

depends on the type of E&M signaling being used. I am refereing to the Cisco router. Correct me if I am wrong, buy when you dial from MICS to Cisco extenstion via E&M life is good? But dialing the other way around it is not working correct?

"The E&M/Direct Inward System Access (DISA) trunk cartridge provides E&M tie line private networking capability between a Norstar and other systems in a private network configuration. It is also equipped with 2 DTMF receivers required for DISA access on Loop Start Trunks. This card supports up to two (2) E&M Type II Trunk interfaces. This trunk cartridge is compatible with the Modular 8x24 (Release DR4 or later) and the Modular ICS. It may be inserted into either the Copper Trunk Module (NT5B32FA) or the Fiber Trunk Module (NTBB20FB93). "

I am not using DISA so I guess this doesn't apply

I have to press "8" to get and E&M line then I dial the cisco extension and then it works. Of course, i would llike to avoid even having to dial 8 but I am not sure if the MICS supports dial plans on E&M lines.

The problem is clearly incoming from the router to the MICS, the calls only get routed to the prime line. Which timing should I be changing on the router?

Prior to changing timing, did you try inserting a pause

how do I insert a pause?

Look at bootom of link

http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f5e.shtml

I looked at the bottom of the page can you be more specific? Where do I need to insert a pause?

Post your latest config for router, I will show you where. Also on the MICS side do you have the E&M programmed for AutoAnswer without DISA enabled?

I figured out by using prefix ,22 will insert a pause

http://groups.google.com/groups?hl=en&lr=&threadm=MPG.16c12e6b358eb907989728%40news.charter.net&rnum=2&prev=/groups%3Fq%3Dcisco%2BE%2526M%2Bdelay%2Bpause%26hl%3Den%26lr%3D%26selm%3DMPG.16c12e6b358eb907989728%2540news.charter.net%26rnum%3D2

AutoAnswer Without DISA enabled is true.

http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f5e.shtml#step6

When I look at the cisco site
*Mar  1 03:45:34.923:   digit=1, components=2,
freq_of_first=697, freq_of_second=1209, amp_of_first=16384,
amp_of_second=16384
*Mar  1 03:45:34.923:   digit=0, components=2,
freq_of_first=941, freq_of_second=1336, amp_of_first=16384,
amp_of_second=16384
*Mar  1 03:45:34.923:   digit=0, components=2,
freq_of_first=941, freq_of_second=1336, amp_of_first=16384,
amp_of_second=16384
*Mar  1 03:45:34.923:   digit=0, components=2,
freq_of_first=941, freq_of_second=1336, amp_of_first=16384,
amp_of_second=16384
*Mar  1 03:45:35.727: vtsp_process_dsp_message: MSG_TX_DIALING_DONE
*Mar  1 03:45:35.727: htsp_process_event: [1/0/0, 1.7 , 19]
em_offhook_digit_donehtsp_alerthtsp_alert_notify

I don't see the em_offhook_digit_donehtsp_alerhtsp_alert_notify line under my digits. The following below it my debug.


1d01h: dsp_close_voice_channel: [1/1/1:289] packet_len=8 channel_id=1 packet_id=75
1d01h: dsp_open_voice_channel_20: [1/1/1:289] packet_len=16 channel_id=1 packet_id=74 alaw_ulaw_select=0 associated_signaling_channel=0 time_slot=65535 serial_port=65535
1d01h: dsp_encap_config: [1/1/1:289] packet_len=30 channel_id=1 packet_id=92
    TransportProtocol 2 t_ssrc=0x0 r_ssrc=0x0 t_vpxcc=0x0 r_vpxcc=0x0
    sid_support=1, tse_payload=65535, seq_num=0x245B, redundancy=0
1d01h: dsp_set_playout_delay
1d01h: dsp_set_playout: [1/1/1:289] packet_len=18 channel_id=1 packet_id=76 mode=1 initial=60 min=40 max=200 fax_nom=300
1d01h: dsp_set_playout_config: [1/1/1:289] packet_len=18 channel_id=1 packet_id=76 mode=1 initial=60 min=40 max=200 fax_nom=300
1d01h: dsp_echo_canceler_control: echo_cancel: 1
1d01h: dsp_echo_canceler_control: [1/1/1:289] echo_cancel 1, disable_hpf 0, flags=0x0, threshold=-21
1d01h: dsp_echo_canceler_control: [1/1/1:289] packet_len=14 channel_id=1 packet_id=66 flags=0x0, threshold=-21, suppressor coverage=7
1d01h: dsp_idle_code_det: [1/1/1:289] packet_len=14 channel_id=1 packet_id=116 enable=0, code=0, duration=6000
1d01h: set_gains: FXx/E&M:  msg->message.set_codec_gains.out_gain=65506
1d01h: dsp_set_gains: [1/1/1:289] packet_len=12 channel_id=1 packet_id=91 in_gain=0 out_gain=65506
1d01h: dsp_vad_enable: [1/1/1:289] enable: packet_len=16 channel_id=1 packet_id=78 thresh=-38 vadtime=250 aggressive=0
1d01h: dsp_encap_config: [1/1/1:289] packet_len=30 channel_id=1 packet_id=92
    TransportProtocol 2 t_ssrc=0x0 r_ssrc=0x0 t_vpxcc=0x0 r_vpxcc=0x0
    sid_support=1, tse_payload=65535, seq_num=0x245B, redundancy=0
1d01h: dsp_voice_mode: [1/1/1:289] cdb 828C4088, cdb->codec_params.modem 2, inband_detect flags 0x21
1d01h: map_dtmf_relay_type--digit relay mode: 2
1d01h: dsp_voice_mode: [1/1/1:289] packet_len=24 channel_id=1 packet_id=73 coding_type=1 voice_field_size=160 VAD_flag=0 echo_length=64 comfort_noise=1 inband_detect=33 digit_relay_mode=2 AGC_flag=0dsp_dtmf_mode(VTSP_TONE_DTMF_MODE)

1d01h: dsp_dtmf_mode: [1/1/1:289] packet_len=10 channel_id=1 packet_id=65 dtmf_or_mf=0
1d01h: dsp_cp_tone_off: [1/1/1:289] packet_len=8 channel_id=1 packet_id=71
1d01h: dsp_cp_tone_on: [1/1/1:289] packet_len=38 channel_id=1 packet_id=72 tone_id=4 n_freq=2 freq_of_first=350 freq_of_second=440 amp_of_first=6942 amp_of_second=6942 direction=1 on_time_first=65535 off_time_first=0 on_time_second=0 off_time_second=0
1d01h: dsp_cp_tone_off: [1/1/1:289] packet_len=8 channel_id=1 packet_id=71
1d01h: dsp_cp_tone_on: [1/1/1:289] packet_len=38 channel_id=1 packet_id=72 tone_id=4 n_freq=2 freq_of_first=350 freq_of_second=440 amp_of_first=6942 amp_of_second=6942 direction=1 on_time_first=65535 off_time_first=0 on_time_second=0 off_time_second=0
1d01h: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=2, rtp_timestamp=0xF79EB35C

1d01h: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=2, duration=175
1d01h: dsp_cp_tone_off: [1/1/1:289] packet_len=8 channel_id=1 packet_id=71
1d01h: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=2, rtp_timestamp=0xF79EB35C

1d01h: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=2, duration=155
1d01h: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=2, rtp_timestamp=0xF79EB35C

1d01h: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=2, duration=175
1d01h: dsp_close_voice_channel: [1/0/0 (290)] packet_len=8 channel_id=1 packet_id=75
1d01h: dsp_open_voice_channel_20: [1/0/0 (290)] packet_len=16 channel_id=1 packet_id=74 alaw_ulaw_select=0 associated_signaling_channel=128 time_slot=65535 serial_port=65535
1d01h: dsp_encap_config: [1/0/0 (290)] packet_len=30 channel_id=1 packet_id=92
    TransportProtocol 2 t_ssrc=0x0 r_ssrc=0x0 t_vpxcc=0x0 r_vpxcc=0x0
    sid_support=1, tse_payload=65535, seq_num=0x189A, redundancy=0
1d01h: dsp_set_playout_delay
1d01h: dsp_set_playout: [1/0/0 (290)] packet_len=18 channel_id=1 packet_id=76 mode=1 initial=60 min=40 max=200 fax_nom=300
1d01h: dsp_set_playout_config: [1/0/0 (290)] packet_len=18 channel_id=1 packet_id=76 mode=1 initial=60 min=40 max=200 fax_nom=300
1d01h: dsp_echo_canceler_control: echo_cancel: 1
1d01h: dsp_echo_canceler_control: [1/0/0 (290)] echo_cancel 1, disable_hpf 0, flags=0x0, threshold=-21
1d01h: dsp_echo_canceler_control: [1/0/0 (290)] packet_len=14 channel_id=1 packet_id=66 flags=0x0, threshold=-21, suppressor coverage=7
1d01h: dsp_idle_code_det: [1/0/0 (290)] packet_len=14 channel_id=1 packet_id=116 enable=0, code=0, duration=6000
1d01h: set_gains: FXx/E&M:  msg->message.set_codec_gains.out_gain=0
1d01h: dsp_set_gains: [1/0/0 (290)] packet_len=12 channel_id=1 packet_id=91 in_gain=0 out_gain=0
1d01h: dsp_vad_enable: [1/0/0 (290)] enable: packet_len=16 channel_id=1 packet_id=78 thresh=-38 vadtime=250 aggressive=0dsp_dtmf_mode(VTSP_TONE_DTMF_MODE)

1d01h: dsp_dtmf_mode: [1/0/0 (290)] packet_len=10 channel_id=1 packet_id=65 dtmf_or_mf=0
1d01h: dsp_dtmf_dialing: [1/0/0 (290)] packet_len=48 channel_id=1 packet_id=90 string=222 digits=3, time_on=100, time_off=100
1d01h:  digit=2, components=2, freq_of_first=697, freq_of_second=1336, amp_of_first=16384, amp_of_second=21000
1d01h:  digit=2, components=2, freq_of_first=697, freq_of_second=1336, amp_of_first=16384, amp_of_second=21000
1d01h:  digit=2, components=2, freq_of_first=697, freq_of_second=1336, amp_of_first=16384, amp_of_second=21000
1d01h: vtsp_process_dsp_message: MSG_TX_DIALING_DONE
1d01h: dsp_idle_mode: [1/0/0 (290)] packet_len=8 channel_id=1 packet_id=68
1d01h: dsp_cp_tone_off: [1/1/1:289] packet_len=8 channel_id=1 packet_id=71
1d01h: dsp_idle_mode: [1/1/1:289] packet_len=8 channel_id=1 packet_id=68
1d01h: dsp_encap_config: [1/1/1:289] packet_len=30 channel_id=1 packet_id=92
    TransportProtocol 2 t_ssrc=0x0 r_ssrc=0x0 t_vpxcc=0x0 r_vpxcc=0x0
    sid_support=143, tse_payload=65535, seq_num=0x1C2D, redundancy=0
1d01h: dsp_voice_mode: [1/1/1:289] cdb 828C4088, cdb->codec_params.modem 2, inband_detect flags 0x20
1d01h: map_dtmf_relay_type--digit relay mode: 2
1d01h: dsp_voice_mode: [1/1/1:289] packet_len=24 channel_id=1 packet_id=73 coding_type=1 voice_field_size=160 VAD_flag=0 echo_length=64 comfort_noise=1 inband_detect=32 digit_relay_mode=2 AGC_flag=0
1d01h: dsp_vad_enable: [1/1/1:289] disable: packet_len=8 channel_id=1 packet_id=77
1d01h: dsp_idle_mode: [1/0/0 (290)] packet_len=8 channel_id=1 packet_id=68
1d01h: dsp_encap_config: [1/0/0 (290)] packet_len=30 channel_id=1 packet_id=92
    TransportProtocol 2 t_ssrc=0x0 r_ssrc=0x0 t_vpxcc=0x0 r_vpxcc=0x0
    sid_support=143, tse_payload=65535, seq_num=0xE0C, redundancy=0
1d01h: dsp_voice_mode: [1/0/0 (290)] cdb 828BB24C, cdb->codec_params.modem 2, inband_detect flags 0x20
1d01h: map_dtmf_relay_type--digit relay mode: 2
1d01h: dsp_voice_mode: [1/0/0 (290)] packet_len=24 channel_id=1 packet_id=73 coding_type=1 voice_field_size=160 VAD_flag=0 echo_length=64 comfort_noise=1 inband_detect=32 digit_relay_mode=2 AGC_flag=0
1d01h: dsp_vad_enable: [1/0/0 (290)] disable: packet_len=8 channel_id=1 packet_id=77
1d01h: dsp_cp_tone_off: [1/1/1:289] packet_len=8 channel_id=1 packet_id=71
1d01h: dsp_idle_mode: [1/1/1:289] packet_len=8 channel_id=1 packet_id=68
1d01h: dsp_get_levels: [1/0/0 (290)] packet_len=8 channel_id=1 packet_id=89
1d01h: vtsp_process_dsp_message: MSG_TX_GET_TX_STAT: rtp_timestamp=0x1B28386D
1d01h: dsp_idle_mode: [1/0/0 (290)] packet_len=8 channel_id=1 packet_id=68
1d01h: dsp_cp_tone_off: [1/1/1:289] packet_len=8 channel_id=1 packet_id=71
1d01h: dsp_idle_mode: [1/1/1:289] packet_len=8 channel_id=1 packet_id=68
1d01h: dsp_close_voice_channel: [1/1/1:289] packet_len=8 channel_id=1 packet_id=75
1d01h: dsp_cp_tone_off: [1/0/0 (290)] packet_len=8 channel_id=1 packet_id=71
1d01h: dsp_idle_mode: [1/0/0 (290)] packet_len=8 channel_id=1 packet_id=68
1d01h: dsp_close_voice_channel: [1/0/0 (290)] packet_len=8 channel_id=1 packet_id=75

Here is the config

version 12.2
service timestamps debug uptime
service timestamps log uptime
service password-encryption
!
hostname cscscovoipgw01
!
enable secret 5 $1$L1Om$uZtzG2jtYAPtq1s4PQ0Q7/
!
ip subnet-zero
!
!
!
ip audit notify log
ip audit po max-events 100
!
call rsvp-sync
!
!
!
voice class h323 1
 h225 timeout tcp establish 3
!
!
!
!
!
!
interface Ethernet0/0
 ip address 10.100.0.32 255.255.255.0
 full-duplex
 h323-gateway voip interface
 h323-gateway voip h323-id cscovoipgw01@cometcomputing.com
!
interface Serial0/0
 no ip address
 shutdown
 no fair-queue
!
ip classless
ip http server
!
!
!
voice-port 1/0/0
 cptone CA
 operation 4-wire
 type 2
!
voice-port 1/0/1
 cptone CA
 operation 4-wire
 type 2
!
voice-port 1/1/0
!
voice-port 1/1/1
!
dial-peer cor custom
!
!
!
dial-peer voice 100 voip
 preference 1
 destination-pattern 20.
 voice-class h323 1
 session target ipv4:10.100.0.40
 dtmf-relay h245-alphanumeric
 ip precedence 5
!
dial-peer voice 1 pots
 destination-pattern 22.
 port 1/0/0
 prefix ,22
!
dial-peer voice 2 pots
 destination-pattern 22.
 port 1/0/1
 prefix ,22
!
dial-peer voice 4 pots
 destination-pattern 231
!
dial-peer voice 230 pots
 destination-pattern 230
 port 1/1/0
!
dial-peer voice 231 pots
 destination-pattern 231
 port 1/1/1
!
gateway

cscscovoipgw01#sh voice port 1/0/0

recEive And transMit 1/0/0 Slot is 1, Sub-unit is 0, Port is 0
 Type of VoicePort is E&M
 Operation State is DORMANT
 Administrative State is UP
 The Last Interface Down Failure Cause is Administrative Shutdown
 Description is not set
 Noise Regeneration is enabled
 Non Linear Processing is enabled
 Music On Hold Threshold is Set to -38 dBm
 In Gain is Set to 0 dB
 Out Attenuation is Set to 0 dB
 Echo Cancellation is enabled
 Echo Cancellation NLP mute is disabled
 Echo Cancellation NLP threshold is -21 dB
 Echo Cancel Coverage is set to 8 ms
 Playout-delay Mode is set to default
 Playout-delay Nominal is set to 60 ms
 Playout-delay Maximum is set to 200 ms
 Playout-delay Minimum mode is set to default, value 40 ms
 Connection Mode is normal
 Connection Number is not set
 Initial Time Out is set to 10 s
 Interdigit Time Out is set to 10 s
 Ringing Time Out is set to 180 s
 Companding Type is u-law
 Region Tone is set for CA

 Analog Info Follows:
 Currently processing none
 Maintenance Mode Set to None (not in mtc mode)
 Number of signaling protocol errors are 0
 Impedance is set to 600r Ohm
 Wait Release Time Out is 30 s
 Station name None, Station number None

 Voice card specific Info Follows:
 Signal Type is wink-start
 Operation Type is 4-wire
 E&M Type is 2
 Dial Type is dtmf
 In Seizure is inactive
 Out Seizure is inactive
 Digit Duration Timing is set to 100 ms
 InterDigit Duration Timing is set to 100 ms
 Pulse Rate Timing is set to 10 pulses/second
 InterDigit Pulse Duration Timing is set to 750 ms
 Clear Wait Duration Timing is set to 400 ms
 Wink Wait Duration Timing is set to 200 ms
 Wait Wink Duration Timing is set to 550 ms
 Wink Duration Timing is set to 200 ms
 Delay Start Timing is set to 300 ms
 Delay Duration Timing is set to 2000 ms
 Dial Pulse Min. Delay is set to 140 ms
 Percent Break of Pulse is 60 percent
 Auto Cut-through is disabled
 Dialout Delay is 70 ms

cscscovoipgw01#sh voice port 1/0/1

recEive And transMit 1/0/1 Slot is 1, Sub-unit is 0, Port is 1
 Type of VoicePort is E&M
 Operation State is DORMANT
 Administrative State is UP
 The Last Interface Down Failure Cause is Administrative Shutdown
 Description is not set
 Noise Regeneration is enabled
 Non Linear Processing is enabled
 Music On Hold Threshold is Set to -38 dBm
 In Gain is Set to 0 dB
 Out Attenuation is Set to 0 dB
 Echo Cancellation is enabled
 Echo Cancellation NLP mute is disabled
 Echo Cancellation NLP threshold is -21 dB
 Echo Cancel Coverage is set to 8 ms
 Playout-delay Mode is set to default
 Playout-delay Nominal is set to 60 ms
 Playout-delay Maximum is set to 200 ms
 Playout-delay Minimum mode is set to default, value 40 ms
 Connection Mode is normal
 Connection Number is not set
 Initial Time Out is set to 10 s
 Interdigit Time Out is set to 10 s
 Ringing Time Out is set to 180 s
 Companding Type is u-law
 Region Tone is set for CA

 Analog Info Follows:
 Currently processing none
 Maintenance Mode Set to None (not in mtc mode)
 Number of signaling protocol errors are 29
 Impedance is set to 600r Ohm
 Wait Release Time Out is 30 s
 Station name None, Station number None

 Voice card specific Info Follows:
 Signal Type is wink-start
 Operation Type is 4-wire
 E&M Type is 2
 Dial Type is dtmf
 In Seizure is inactive
 Out Seizure is inactive
 Digit Duration Timing is set to 100 ms
 InterDigit Duration Timing is set to 100 ms
 Pulse Rate Timing is set to 10 pulses/second
 InterDigit Pulse Duration Timing is set to 750 ms
 Clear Wait Duration Timing is set to 400 ms
 Wink Wait Duration Timing is set to 200 ms
 Wait Wink Duration Timing is set to 550 ms
 Wink Duration Timing is set to 200 ms
 Delay Start Timing is set to 300 ms
 Delay Duration Timing is set to 2000 ms
 Dial Pulse Min. Delay is set to 140 ms
 Percent Break of Pulse is 60 percent
 Auto Cut-through is disabled
 Dialout Delay is 70 ms

Here is something interesting to note:

On my norstar system, when I sieze the trunk line and try to call back into an extension, (pressing 8) it always goes back to the prime line. I think this is the cause of the problem, I should be able to dial "8" from the nortel MICS and then press and internal extension number like 222 and it should route to the phone. It is not doing that right now, it always routes back to the prime line 221.

Disregard the last post, that my wires were still plugged into the cisco router. What I need to do is have a cable made that will loop back E&M port 1 to E&M port 2 on the MICS. How do I make this cable?

How do you have the E&M cards discharging from the system? Are you using and Amphenol cable now? What type of cable do you have between the MICS and Cisco? You would want a crossover cable.

1-8
2-7
3-6
4-5

Yes I am using the Amphenol cable right now. I just want to have it so that the MICS calls back to the MICS using the E&M trunk. This would be a good test do you agree? How should I make the cables?

I read that I could just use the cisco console cable to do this. Also do i have to configure a dial plan in the MICS under system programming? We are so close!....

Yes I agree on the E&M check my post in this thread Dated: 11/30/2004 12:43PM PST (LOL)

Looked at your config, just add the comma after prefix: no need to add the 22 (it would add two more "2"'s)


If you want transparent routing you would need to config a dial plan in the MICS, this can be domne using Routing Services under the Services part of programming. In orderto to that I would recommend using extentiosn numbers that are diffrent between the systems, i.e 200 serie for MICS, 300 sereis for Cisco phones.

How does the Amphenol cable terminate right now? What is connected to the end of it?

I just terminate it into a regular RJ45 patch panel so that it is easier to plug and unplug. All the pin assignments work when plugging to the cisco e&m port. I am guessing I need to match pin1 or port 1 to pin 8 or port 2?

Port 1 to the amphenol cable is line 1
Port 2 to the amphenol cable is line 2

The wiring connections match those from the cisco site for pin 1 to pin 8 which is how I wired them so I could use a straight cable to plug directlry into the E&M port of my cisco router.

 

Currently when I use a straight or a rollover console cable and try to dial nothing works. I am guessing the problem is on the MICS side.

I used a cisco console rollovercable and when I dial, it seems to seize the E&M trunk but there is no dialtone. When I dial 222, nothing happens, and it doesn't forward the extension internally. Wiring most likely?

The patch panel you are using is it wired for 568A or 568B?

it is just a regular rj45 cat 5 e patch panel

I got it working with dialtone now,it was a wiring config, I did a cross connect. But when I dial the digits, it isn't picking up the digits. For example when I press 222, i still only get dialtone on the MICS

Nothing regular about a patch panel, do not take this the wrong way, but I find it amazing after all these years that 90% of IT pro's do not understand the cabling that connects it all together. Needless to say that was Cisco's biggest complaint about the CCIE 2 day test, Day 1, had to cable everything, now they made it a one day test and pre-wired the setup.

Any hoot, off my soapbox, make and model of patch panel? Also need to see if its wired for 568A or 568B, difference is that the white/orange and white/green pairs are swapped betwen the two wiring specs.

OKAY! Got the wiring and everything working, confirm it is the PBX, since when I call back the E&M line it automaticallly goes to the Prime number. So it is clearly an MICS setting, I press 8 and then it gives me dial tone and when I dial 222 it defaults to 221, so there must be a system setting somewhere.

Hmmmm....dial from 221 and see what happens?

same thing it goes straight to the prime line. I have assigned the 2 E&M lines to Line pool B

okay give me a few minutes to figure it out.

Before we dive into it, is DND enabled on ext 222 ?

I don't think so because when I call it regulary without using the E&M Line, it picks up.

What is the dial mode of the E&M line, needs to be set to tone, the deafult is pulse which could explain the results.

Also under programming for the E&M line is it assigend as

                              private to: X

What model phone?

Both E&M Lines are assigned to POOL B which require you to dial 8 to get out. Neither of them are assigned as private lines. The model phone I am using is a T7316E.

E&M lines set for Tone on dial mode? Answer mode is set to manual? ANI and DNIS settings are set to yes?

Also do you have a buttset?

You could palce your buttset on the ausio pairs and verify digits are being passed.

Also make sure singaling is winkstart not imeadiate

Also make sure that the lines are not set as target lines.

no buttset, will go buy one tommorrow, tried manual ANI and DNIS, none work. When I try to use ANI and DNIS, it craps out.

signaling is winkstart.

Verify that its not a target line, also check that under teminals and sets that ext222 has been giving access to the pool and the lines are set to appear and ring.

My toy store:

http://www.specialized.net/ecommerce/shop/frameset.htm    good starter would be Ranger-Lil' Buttie w/ ADLI

I set the set 222 to appear and ring and it does but only when I set the answer mode to manual. Also, As soon as I press 8 to seize the trunk the phone rings before I even dial the extension 222 when the answer mode is set to auto. When I set the answer mode to auto, it defaults back to the prime line 221.

Seems like that the trunk has been assiged to that ext directly as private or target line. Does ext 221 have the same settings as 222 ?

i am starting all over again this should make things easier, let me get back to you. I did notice that one of the trunk lines had a Private on it but the other didn't not sure how that go there.

Great minds think alike, just was thinking, was the system defaulted when we started all of this?

yes, I also changed the starting extensions to 101 to match those of our system in real use right now.

no such luck, same problem, it always defaults back to the prime line, guess I will call Nortel tommorrow... wife is going to be so mad...staying so late!

Harry,

Have you been able to resolve this?

NOPE, went home to bed, i haven't even looked today, is there a good site for reading about this for the norstar?

Not that I know of off hand. When you programmed the FXS lines instead of the E&M you did not have any issues correct?

Also did you actually default the entire software back to original factory settings?

same issues, when I had the fxs lines call to the norstar MICS, it still automatically goes to the prime line.

Yup defaulted EVERYTHING