Cisco 7942 DTMF issue on outgoing calls

Asterisk Asterisk Support
1

This is a cross post to a posting I put on the Cisco forums, but got no traction with… Maybe the team here has seen the issue.

I just bought a Cisco 7942 phone to connect to my asterisk PBX. I have installed the latest SIP firmware, and put together a SEP.cnf.xml. I’m able to make phone calls at this point. When I try to use the keypad to select options on a remote call system, it doesn’t work. On most phones, I’m able to set the DTMF mode to be rfc2833 compliant and it works fine, however there seems to be a lack of documentation on the possible settings that can go in the SEP.cnf.xml files.

My SEP.cnf.xml file and sip.conf file for asterisk are below. Two questions:

  1. Where can I go for details on SEP.cnf.xml files (and the other needed files). The asterisk, voip-info try hard to define the different parameters, but all the options aren’t fully doucmented. A good document to configure the phone would be great.
  2. What are the correct DTMF settings that I should be using for this?
  3. Any other suggestions for my configuration? Security wise, sanity wise, etc.

Thanks,
Scott

Relevent sip.conf:

[quote][general]
context=default_context
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp
srvlookup=yes
dynamic_exclude_static = yes
buggymwi=yes
contactpermit=192.168.1.0/24

[laura_office]
type=friend
host=dynamic
context=home
username=laura_office
secret=************
mailbox=100@home
subscribemwi=yes
dtmfmode=info
disallow=all
allow=ulaw
insecure=invite
; deny=0.0.0.0/0.0.0.0
; permit=192.168.1.0/255.255.255.0[/quote]

SEP.cnf.xml

[quote]
SIP
username
************


M-D-YA
Eastern Standard/Daylight Time


192.168.1.1
Unicast








2000
5060
5061

192.168.1.15







true


true
x–serviceuri-cfwdall
x-cisco-serviceuri-pickup
x-cisco-serviceuri-opickup
x-cisco-serviceuri-gpickup
x-cisco-serviceuri-meetme
x-cisco-serviceuri-abbrdial
false
2
true
true
2
2
0
true


6
10
180
3600
5
120
120
5
500
4000
70
false
None

1
false
true
false
false
101
3
avt
false
false
3
Laura Office
false
3
false
16384
32766


9
Laura Office
USECALLMANAGER
5060
laura_office
Laura Office

2

3
laura_office
***************
false
1
9999
4
5
laura_office

true
false
false
true



dialplan.xml


false
false
1
1
0
0
0
0
0
22
0
1
1





en_US







1.0.0.0-1

1
1
SIP42.9-3-1-1S

[/quote]

2

I’ve had a problem when using SIP info messages to transmit DTMF.

Have you configured the phone to send SIP info message for DTMF?

The RFC specifies to use uppercase characters for the DTMF A to D. some SIP phones (Linksys for example) use lowercase characters. Asterisk followed the RFC to the letter (ha ha) and only looked for uppercase A to D.

Can you turn on SIP debug and see what your phone is transmitting for the DTMF?

If you are seeing lowercase characters in the SIP info messages you can patch chan_sip.c where it is looking for only uppercase.
method: handle_request_info
add
else if ((buf[0] >= ‘a’) && (buf[0] <= ‘d’))
event = 12 + buf[0] - ‘a’;

method: add_digit
add
else if ((digit >= ‘a’) && (digit <= ‘d’))
event = 12 + digit - ‘a’;