VOIP and E&M
Mr Lightfoot
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.
Steven A. Ridder
of wink-start (if you are using wink-start).
"Mr Lightfoot" <fob...@bbnplanet.net> wrote in message
news:Xns91A062107...@24.9.139.141...
Dave Phelps
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?
In article <Xns91A062107...@24.9.139.141>, fob...@bbnplanet.net
says...
--
Dave Phelps
Phone Masters Ltd.
deadspam=tippenring
Karate-Kid
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 ?
"Mr Lightfoot" <fob...@bbnplanet.net> wrote in message
news:Xns91A062107...@24.9.139.141...
Mr Lightfoot
news:MPG.16ba8f619...@news.charter.net:
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
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.
In article <Xns91A4AC1B0...@24.9.139.141>, fob...@bbnplanet.net
says...
--
swamp thing
news:MPG.16c12e6b3...@news.charter.net:
>> 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.