VOIP and E&M

VOIP and E&M

Post by Mr Lightfoo » Fri, 25 Jan 2002 17:35:18



We're in the process of configuring a pair of routers with E&M modules to
connect two offices.  We are able to dial through the PBX, to the E&M
module on router 1, over the F/R to router 2, but all we get is dead air at
that point.

The same thing happens if we go the other way from the router 2 side.  We
get all the way through then dead air.  Looking at the call status, it says
it was sucessful. Does the last leg E&M pass the called extension numbers
as tones to the receiving PBX?   Should the receiving PBX be providing a
dial tone?  Thanks for any help.

 
 
 

VOIP and E&M

Post by Steven A. Ridde » Fri, 25 Jan 2002 20:13:21


try dtmf relay in dial-peer.  also, try immediate forward signalling instead
of wink-start (if you are using wink-start).


Quote:> We're in the process of configuring a pair of routers with E&M modules to
> connect two offices.  We are able to dial through the PBX, to the E&M
> module on router 1, over the F/R to router 2, but all we get is dead air
at
> that point.

> The same thing happens if we go the other way from the router 2 side.  We
> get all the way through then dead air.  Looking at the call status, it
says
> it was sucessful. Does the last leg E&M pass the called extension numbers
> as tones to the receiving PBX?   Should the receiving PBX be providing a
> dial tone?  Thanks for any help.


 
 
 

VOIP and E&M

Post by Dave Phelp » Sat, 26 Jan 2002 12:39:14


What type of PBX are you using?

Do you have your E&M trunks pgm'd to provide dial tone? Do you need to
assign DTMF receivers to your E&M trunks?

Yes, the routers will pass digits to the pbx if you're not absorbing
them. Can you post your dial-peer and voice-port configs?


says...

Quote:> We're in the process of configuring a pair of routers with E&M modules to
> connect two offices.  We are able to dial through the PBX, to the E&M
> module on router 1, over the F/R to router 2, but all we get is dead air at
> that point.

> The same thing happens if we go the other way from the router 2 side.  We
> get all the way through then dead air.  Looking at the call status, it says
> it was sucessful. Does the last leg E&M pass the called extension numbers
> as tones to the receiving PBX?   Should the receiving PBX be providing a
> dial tone?  Thanks for any help.

--
Dave Phelps
Phone Masters Ltd.
deadspam=tippenring
 
 
 

VOIP and E&M

Post by Karate-Ki » Sat, 26 Jan 2002 23:28:43


Does the remote phone ring (you will not hear the ring tone) ?

If it does then you've got your signalling (e and m) wires correct but
you've got the speech pairs wrong.  I assume you are using 4-wire (you
should be in order to prevent echo).

Yo have got the signalling type correct such as immediate start or wink
start ?  Cisco's web site has some great documents on this - just search
under E+M.  Is your PBX a Siemens HiCom with the new type of E+M card ?


Quote:> We're in the process of configuring a pair of routers with E&M modules to
> connect two offices.  We are able to dial through the PBX, to the E&M
> module on router 1, over the F/R to router 2, but all we get is dead air
at
> that point.

> The same thing happens if we go the other way from the router 2 side.  We
> get all the way through then dead air.  Looking at the call status, it
says
> it was sucessful. Does the last leg E&M pass the called extension numbers
> as tones to the receiving PBX?   Should the receiving PBX be providing a
> dial tone?  Thanks for any help.

 
 
 

VOIP and E&M

Post by Mr Lightfoo » Wed, 30 Jan 2002 09:55:15




> What type of PBX are you using?

> Do you have your E&M trunks pgm'd to provide dial tone? Do you need to
> assign DTMF receivers to your E&M trunks?

> Yes, the routers will pass digits to the pbx if you're not absorbing
> them. Can you post your dial-peer and voice-port configs?



>> We're in the process of configuring a pair of routers with E&M modules
>> to connect two offices.  We are able to dial through the PBX, to the
>> E&M module on router 1, over the F/R to router 2, but all we get is
>> dead air at that point.

>> The same thing happens if we go the other way from the router 2 side.
>> We get all the way through then dead air.  Looking at the call status,
>> it says it was sucessful. Does the last leg E&M pass the called
>> extension numbers as tones to the receiving PBX?   Should the
>> receiving PBX be providing a dial tone?  Thanks for any help.

We're using Mitel ML200L's at both ends.  

I'm not sure about the E&M trunks.  

voice-port 2/0
 timeouts wait-release 3
 connection trunk +19197741000
 description connected to PBX1 (303555-xxxx)
!
voice-port 2/1
 timeouts wait-release 3
 connection trunk +19197741000
 description connected to PBX1 (303555-xxxx)
!
dial-peer voice 1 pots
 destination-pattern +13035551000
 port 2/0
!
dial-peer voice 2 pots
 destination-pattern +13035551000
 port 2/1
!
dial-peer voice 3 voip
 destination-pattern +19197741000
 session target ipv4:10.191.4.2
 ip precedence 5
!
dial-peer voice 4 voip
 destination-pattern +19197741000
 session target ipv4:10.191.4.2
 ip precedence 5

 sh voice port 2/0

 recEive And transMit 2/0 Slot is 0, Sub-unit is 2, Port is 0
 Type of VoicePort is E&M
 Operation State is DOWN
 Administrative State is UP
 The Last Interface Down Failure Cause is Administrative Shutdown
 Description is connected to AnaPBX (714555-xxxx)
 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 Cancel Coverage is set to 8 ms
 Connection Mode is trunk
 Connection Number is +19097741000
 Initial Time Out is set to 10 s
 Interdigit Time Out is set to 10 s
 Call-Disconnect Time Out is set to 60 s
 Ringing Time Out is set to 180 s
 Companding Type is u-law
 Region Tone is set for US

 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 3 s
 Station name None, Station number None

 Voice card specific Info Follows:
 Signal Type is wink-start
 Operation Type is 2-wire
 E&M Type is 1
 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 for immediate start is 300 ms

sh voic trunk sup 2/0
2/0 : state : TRUNK_SC_OOS_SND_OOS, voice : off, signal : off, master
        status: trunk disconn
        sequence oos : idle and oos
        pattern :rx_idle = 0000 rx_oos = 1111
        timing : idle = 0, restart = 120, standby = 0, timeout = 30
        supp_all = 0, supp_voice = 0, keep_alive = 5
        timer: oos_ais_timer = 0, timer = 322
Anaheim#sh voic trunk sup 2/1
2/1 : state : TRUNK_SC_OOS_SND_OOS, voice : off, signal : off, master
        status: trunk disconn
        sequence oos : idle and oos
        pattern :rx_idle = 0000 rx_oos = 1111
        timing : idle = 0, restart = 120, standby = 0, timeout = 30
        supp_all = 0, supp_voice = 0, keep_alive = 5
        timer: oos_ais_timer = 0, timer = 323                          

sh voice dsp
                                BOOT                      PAK
TYPE DSP CH CODEC    VERS STATE STATE   RST AI PORT    TS ABORT   TX/RX-
PAK-CNT
==== === == ======== ==== ===== ======= === == ======= == =====
===============
DSP# 0: state IN SERVICE, 2 channels allocated
 channel# 0: voice port 2/0, codec g729r8, state UP
 channel# 1: voice port 2/1, codec g729r8, state UP                                

sh voice trac 2/1
2/1 State Transitions: (state, event) -> (state, event) ...
(S_OPEN_PEND, E_DSP_INTERFACE_INFO) -> (S_DOWN, E_HTSP_IF_INSERVICE) ->
(S_OPEN_PEND, E_DSP_INTERFACE_INFO) -> (S_DOWN, E_HTSP_IF_INSERVICE) ->
(S_OPEN_PEND, E_HTSP_GO_TRUNK) -> (S_UP, E_HTSP_IF_OOS) ->
(S_UP, E_HTSP_EVENT_TIMER) -> (S_UP, UNKNOWN_HTSP_EVENT) ->
(S_UP, E_HTSP_RELEASE_REQ) -> (S_UP, E_HTSP_IF_OOS_CONF) ->
(S_OPEN_PEND, E_HTSP_IF_INSERVICE) ->                                  

 
 
 

VOIP and E&M

Post by Dave Phelp » Thu, 31 Jan 2002 14:46:23


Quote:> dial-peer voice 1 pots
>  destination-pattern +13035551000
>  port 2/0

Your destination patterns are absorbing all your digits. No digits are
being sent to your remote PBX.

Here's one of mine...

dial-peer voice 2300 pots
 destination-pattern 23..
 port 2/0
 prefix ,23
!
dial-peer voice 2301 pots
 destination-pattern 23..
 port 2/1
 prefix ,23

Notice that the digits absorbed by the destination-pattern (23), I'm
reinserting on the trunk. The ',' is there to insert a pause, because the
Cisco will send the digits to Nortel Norstars before the Norstar has
assigned a receiver, causing the first digit to be missed (took me
forever to figure that out). The 2 digits represented by the '..' are
passed to the PBX after the prefix.

If this is your problem, you can place a call to the remote switch, then
when you get silence, dial the digits (on the phone you are using) that
the remote switch is expecting and see if your call gets sent to the
correct destination.


says...



> > What type of PBX are you using?

> > Do you have your E&M trunks pgm'd to provide dial tone? Do you need to
> > assign DTMF receivers to your E&M trunks?

> > Yes, the routers will pass digits to the pbx if you're not absorbing
> > them. Can you post your dial-peer and voice-port configs?



> >> We're in the process of configuring a pair of routers with E&M modules
> >> to connect two offices.  We are able to dial through the PBX, to the
> >> E&M module on router 1, over the F/R to router 2, but all we get is
> >> dead air at that point.

> >> The same thing happens if we go the other way from the router 2 side.
> >> We get all the way through then dead air.  Looking at the call status,
> >> it says it was sucessful. Does the last leg E&M pass the called
> >> extension numbers as tones to the receiving PBX?   Should the
> >> receiving PBX be providing a dial tone?  Thanks for any help.

> We're using Mitel ML200L's at both ends.  

> I'm not sure about the E&M trunks.  

> voice-port 2/0
>  timeouts wait-release 3
>  connection trunk +19197741000
>  description connected to PBX1 (303555-xxxx)
> !
> voice-port 2/1
>  timeouts wait-release 3
>  connection trunk +19197741000
>  description connected to PBX1 (303555-xxxx)
> !
> dial-peer voice 1 pots
>  destination-pattern +13035551000
>  port 2/0
> !
> dial-peer voice 2 pots
>  destination-pattern +13035551000
>  port 2/1
> !
> dial-peer voice 3 voip
>  destination-pattern +19197741000
>  session target ipv4:10.191.4.2
>  ip precedence 5
> !
> dial-peer voice 4 voip
>  destination-pattern +19197741000
>  session target ipv4:10.191.4.2
>  ip precedence 5

>  sh voice port 2/0

>  recEive And transMit 2/0 Slot is 0, Sub-unit is 2, Port is 0
>  Type of VoicePort is E&M
>  Operation State is DOWN
>  Administrative State is UP
>  The Last Interface Down Failure Cause is Administrative Shutdown
>  Description is connected to AnaPBX (714555-xxxx)
>  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 Cancel Coverage is set to 8 ms
>  Connection Mode is trunk
>  Connection Number is +19097741000
>  Initial Time Out is set to 10 s
>  Interdigit Time Out is set to 10 s
>  Call-Disconnect Time Out is set to 60 s
>  Ringing Time Out is set to 180 s
>  Companding Type is u-law
>  Region Tone is set for US

>  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 3 s
>  Station name None, Station number None

>  Voice card specific Info Follows:
>  Signal Type is wink-start
>  Operation Type is 2-wire
>  E&M Type is 1
>  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 for immediate start is 300 ms

> sh voic trunk sup 2/0
> 2/0 : state : TRUNK_SC_OOS_SND_OOS, voice : off, signal : off, master
>         status: trunk disconn
>         sequence oos : idle and oos
>         pattern :rx_idle = 0000 rx_oos = 1111
>         timing : idle = 0, restart = 120, standby = 0, timeout = 30
>         supp_all = 0, supp_voice = 0, keep_alive = 5
>         timer: oos_ais_timer = 0, timer = 322
> Anaheim#sh voic trunk sup 2/1
> 2/1 : state : TRUNK_SC_OOS_SND_OOS, voice : off, signal : off, master
>         status: trunk disconn
>         sequence oos : idle and oos
>         pattern :rx_idle = 0000 rx_oos = 1111
>         timing : idle = 0, restart = 120, standby = 0, timeout = 30
>         supp_all = 0, supp_voice = 0, keep_alive = 5
>         timer: oos_ais_timer = 0, timer = 323                          

> sh voice dsp
>                                 BOOT                      PAK
> TYPE DSP CH CODEC    VERS STATE STATE   RST AI PORT    TS ABORT   TX/RX-
> PAK-CNT
> ==== === == ======== ==== ===== ======= === == ======= == =====
> ===============
> DSP# 0: state IN SERVICE, 2 channels allocated
>  channel# 0: voice port 2/0, codec g729r8, state UP
>  channel# 1: voice port 2/1, codec g729r8, state UP                                

> sh voice trac 2/1
> 2/1 State Transitions: (state, event) -> (state, event) ...
> (S_OPEN_PEND, E_DSP_INTERFACE_INFO) -> (S_DOWN, E_HTSP_IF_INSERVICE) ->
> (S_OPEN_PEND, E_DSP_INTERFACE_INFO) -> (S_DOWN, E_HTSP_IF_INSERVICE) ->
> (S_OPEN_PEND, E_HTSP_GO_TRUNK) -> (S_UP, E_HTSP_IF_OOS) ->
> (S_UP, E_HTSP_EVENT_TIMER) -> (S_UP, UNKNOWN_HTSP_EVENT) ->
> (S_UP, E_HTSP_RELEASE_REQ) -> (S_UP, E_HTSP_IF_OOS_CONF) ->
> (S_OPEN_PEND, E_HTSP_IF_INSERVICE) ->                                  

--
Dave Phelps
Phone Masters Ltd.
deadspam=tippenring
 
 
 

VOIP and E&M

Post by swamp thin » Fri, 01 Feb 2002 03:36:57




Quote:>> dial-peer voice 1 pots
>>  destination-pattern +13035551000 port 2/0
> Your destination patterns are absorbing all your digits. No digits are
> being sent to your remote PBX.

> Here's one of mine...

> dial-peer voice 2300 pots
>  destination-pattern 23..
>  port 2/0
>  prefix ,23
> !
> dial-peer voice 2301 pots
>  destination-pattern 23..
>  port 2/1
>  prefix ,23

> Notice that the digits absorbed by the destination-pattern (23), I'm
> reinserting on the trunk. The ',' is there to insert a pause, because
> the Cisco will send the digits to Nortel Norstars before the Norstar
> has assigned a receiver, causing the first digit to be missed (took me
> forever to figure that out). The 2 digits represented by the '..' are
> passed to the PBX after the prefix.

> If this is your problem, you can place a call to the remote switch,
> then when you get silence, dial the digits (on the phone you are using)
> that the remote switch is expecting and see if your call gets sent to
> the correct destination.

<snip>

Okay, I'll give this a try.  I think we did try dialing some digits after
one of the connects and nothing happened.  

Thanks.

 
 
 

1. VoIP, E&M and nailed up audio lines

A possible application has arisen for linking a pair of voice intercomm
systems at a couple of sites.  These intercomms expect to be joined
together by a standard telco 4 wire leased line.  There is no
signalling component at all, just audio.

The cisco MC3810 with E&M interfaces can present a suitable 4 wire
audio interface.  A search on the cisco site produces the following
command:

# voice-port slot/port
#connection {plar | tie-line | trunk | plar-opx} destination-string
[answer-mode]

Which suggests what is required can be set up.

Has anyone actually done this and if so, does it work OK?

Also, when the IP line fails and then comes back up, does the audio
line re-establish itself, or is this something that needs to be done
manually?

Thanks in advance.

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