[Asterisk-Users] Re: 0.7.2 with cisco router 7960

2004-03-26 Thread Daniel Cubero Salas, Ing
Our cisco router have these dial peers: 

dial-peer voice 900 pots
application session
destination-pattern 5000
port 1/0/0
!
dial-peer voice 800 pots
application session
destination-pattern 9
port 1/1/1
!
dial-peer voice 701 pots
application session
destination-pattern 3003
port 1/0/1
!
dial-peer voice 10 pots
application session
destination-pattern 13T
port 0/0:1-- Channelized E1
!
dial-peer voice 5 pots
incoming called-number X00
direct-inward-dial
!
dial-peer voice 35 pots
application session
destination-pattern 12T
port 1/1/1
!
dial-peer voice 36 pots
application session
destination-pattern 14T
port 1/1/0
!
dial-peer voice 1 voip
application session
destination-pattern ...
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 4 pots
incoming called-number X10
direct-inward-dial
!
dial-peer voice 6 pots
incoming called-number X11
direct-inward-dial
!
dial-peer voice 7 pots
incoming called-number X12
direct-inward-dial
! 

=
When the call is from PSTN, detection of DTMF by Asterisk+Cisco 2600 works 
pretty well; but when the call is from Cisco 7960 phone thru ASTERISK+Cisco 
2600 to PSTN (like IVR o PBX) always DTMF tones (for long number example 4 
or more) aren´t recognized or it has wrong detection (I digit 9228373 but 
PBX in PSTN seen 928373 or 9287 or 922283). 

am I missing anything? 

Regards 

Daniel 

Pd. What is meaning of CME? 



Kurt Pasewaldt writes: 

What does your VoIP dial peer look like?
Does it have dtmf-relay rtp-nte under the VoIP
dial peer.  This will enable RC2833.  This assume you 
are not running the CME load on the router. 

Kurt 

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[Asterisk-Users] Re: 0.7.2 with cisco router 7960

2004-03-26 Thread Daniel Cubero Salas, Ing
yes, the 7960 is sending the right digits, because in message log from 
asterisk I can see each dtmf. A brief message log is below: 

Mar 25 19:28:33 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 
(SIP/2010-9164)
Mar 25 19:28:33 DEBUG[1200826048]: Bridge stops bridging channels 
SIP/2010-9164 and SIP/cisco2600-2b14
Mar 25 19:28:33 DEBUG[1209214528]: Difference is 976, ms is 142
Mar 25 19:28:33 DEBUG[1209214528]: Auto-deactivating generator
Mar 25 19:28:33 DEBUG[1200826048]: Difference is 3192, ms is 419
Mar 25 19:28:33 DEBUG[1200826048]: Difference is 4280, ms is 555
Mar 25 19:28:38 DEBUG[1200826048]: Sending dtmf: 57 (9)
Mar 25 19:28:38 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 
(SIP/2010-9164)
Mar 25 19:28:38 DEBUG[1200826048]: Bridge stops bridging channels 
SIP/2010-9164 and SIP/cisco2600-2b14
Mar 25 19:28:38 DEBUG[1209214528]: Auto-deactivating generator
Mar 25 19:28:39 DEBUG[1200826048]: Difference is 3336, ms is 437
Mar 25 19:28:39 DEBUG[1200826048]: Difference is 4296, ms is 557
Mar 25 19:28:39 DEBUG[1200826048]: Sending dtmf: 50 (2)
Mar 25 19:28:39 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 
(SIP/2010-9164)
Mar 25 19:28:39 DEBUG[1200826048]: Bridge stops bridging channels 
SIP/2010-9164 and SIP/cisco2600-2b14
Mar 25 19:28:39 DEBUG[1209214528]: Auto-deactivating generator
Mar 25 19:28:40 DEBUG[1200826048]: Difference is 3336, ms is 437
Mar 25 19:28:40 DEBUG[1200826048]: Difference is 4296, ms is 557
Mar 25 19:28:40 DEBUG[1200826048]: Sending dtmf: 50 (2)
Mar 25 19:28:40 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 
(SIP/2010-9164)
Mar 25 19:28:40 DEBUG[1200826048]: Bridge stops bridging channels 
SIP/2010-9164 and SIP/cisco2600-2b14
Mar 25 19:28:40 DEBUG[1209214528]: Difference is 2104, ms is 283
Mar 25 19:28:40 DEBUG[1209214528]: Auto-deactivating generator
Mar 25 19:28:40 DEBUG[1200826048]: Difference is 2072, ms is 279
Mar 25 19:28:40 DEBUG[1200826048]: Difference is 4288, ms is 556
Mar 25 19:28:41 DEBUG[1200826048]: Sending dtmf: 56 (8)
Mar 25 19:28:41 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 
(SIP/2010-9164)
Mar 25 19:28:41 DEBUG[1200826048]: Bridge stops bridging channels 
SIP/2010-9164 and SIP/cisco2600-2b14
Mar 25 19:28:41 DEBUG[1209214528]: Difference is 824, ms is 123
Mar 25 19:28:41 DEBUG[1209214528]: Auto-deactivating generator
Mar 25 19:28:41 DEBUG[1200826048]: Difference is 3336, ms is 437
Mar 25 19:28:41 DEBUG[1200826048]: Difference is 4296, ms is 557
Mar 25 19:28:42 DEBUG[1200826048]: Sending dtmf: 51 (3)
Mar 25 19:28:42 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 
(SIP/2010-9164)
Mar 25 19:28:42 DEBUG[1200826048]: Bridge stops bridging channels 
SIP/2010-9164 and SIP/cisco2600-2b14
Mar 25 19:28:42 DEBUG[1209214528]: Difference is 1144, ms is 163
Mar 25 19:28:42 DEBUG[1209214528]: Auto-deactivating generator
Mar 25 19:28:42 DEBUG[1200826048]: Difference is 3032, ms is 399
Mar 25 19:28:42 DEBUG[1200826048]: Difference is 4296, ms is 557
Mar 25 19:28:43 DEBUG[1200826048]: Sending dtmf: 55 (7)
Mar 25 19:28:43 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 
(SIP/2010-9164) 

I tought than the wrong interpretation or transport is on Cisco 2600 when 
the call is outgoing and use DTMF (the voice is sending without trouble) 

Daniel 

Kurt Pasewaldt writes: 

Daniel, 

Can you determine if the 7960 is sending the right
amount of digits. 

CME = Cisco Call Manager Express  (PBX)
Its a scaled down version of Call Manage and it can be
ran on the following routers: 

1751-v
1760 1760-v
2610XM
2611XM
2620XM
2650XM
2651XM-V
2691
3640 3640-A
3660
3725/45
IAD2420 

--- Daniel Cubero Salas, Ing [EMAIL PROTECTED]
wrote:
Our cisco router have these dial peers:  

dial-peer voice 900 pots
application session
destination-pattern 5000
port 1/0/0
!
dial-peer voice 800 pots
application session
destination-pattern 9
port 1/1/1
!
dial-peer voice 701 pots
application session
destination-pattern 3003
port 1/0/1
!
dial-peer voice 10 pots
application session
destination-pattern 13T
port 0/0:1-- Channelized E1
!
dial-peer voice 5 pots
incoming called-number X00
direct-inward-dial
!
dial-peer voice 35 pots
application session
destination-pattern 12T
port 1/1/1
!
dial-peer voice 36 pots
application session
destination-pattern 14T
port 1/1/0
!
dial-peer voice 1 voip
application session
destination-pattern ...
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 4 pots
incoming called-number X10
direct-inward-dial
!
dial-peer voice 6 pots
incoming called-number X11
direct-inward-dial
!
dial-peer voice 7 pots
incoming called-number X12
direct-inward-dial
!  

=
When the call is from PSTN, detection of DTMF by
Asterisk+Cisco 2600 works 
pretty well; but when the call is from Cisco 7960
phone thru ASTERISK+Cisco 
2600 to PSTN (like IVR o PBX) always DTMF tones (for
long number example 4 
or more) aren´t recognized or it has wrong detection
(I digit 9228373