[Asterisk-Users] audio problems between asterisk and Cisco 7910 using SCCP

[Mark Mills]

Hi,

I am working with a friend to setup two Asterisk servers over the 
internet, one at each location and using IAX2 for trunking calls, using 
Cisco 7910 phones and chan_sccp.   The phones are all the same hardware 
and firmware revisions.

Lets call the sites AsteriskA and AsteriskB.   PhoneA is at AsteriskA, 
PhoneB is at AsteriskB.

PhoneA has problems, when calling the local voice mail service at
AsteriskA, the prompts are heard, button presses work, but audio does not
appear to reach the asterisk server.  The following error message appears
within the asterisk console:

Jun  1 08:43:01 WARNING[13326]: app_voicemail.c:1222 play_and_record: No 
audio available on SCCP/201-00000001

The voice mail files that are created are empty.   Performing a packet 
dump I do see packets going to the Asterisk server.

Now also IAX2 is setup between AsteriskA and AsteriskB, and that seems to 
be functioning.   PhoneA and PhoneB can call each other from either 
direction, but once again there is no sound coming from PhoneA, its only 
one way.   If PhoneA is not answered, voicemail works and PhoneB can leave 
messages that PhoneA can retrieve, but not the other way around.

We performed a packet dump When making calls between the two locations, 
PhoneA sends data to AsteriskA, but AsteriskA doesnt forward it to 
AsteriskB.   It seems that the voice traffic is going from PhoneA is not 
being accepted at all?

Below is the config files that are in use for this setup. This has been 
compiled from source using asterisk-0.9.0.tar.gz and 
chan_sccp.02-easter.tar, on a Redhat 9 box running kernel 2.4.20-8.     

Does anyone have any idea what could be the problem and what we have 
missed?

Thanks,
  Mark


/etc/asterisk/sccp.conf
==========================
[general]

keepalive = 300
context = default
dateFormat = D/M/Y  

[SEP000427E8CD80]
type            = 7910
autologin       = 201
description     = Extension 201

[201]
id              = 201
pin             = 1234
label           = Mark Mills <201>
description     = Mark Cisco 7910 Phone
callwaiting     = 1
mailbox         = 201
callerid        = "Mark Mills", <201>




/etc/asterisk/extensions.conf
==========================
[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for 
demo

[unknown]

exten => _.,1,Congestion

[default]

exten => 201,1,Macro(std-exten,SCCP/201,40)
exten => _1XX,1,Dial(IAX2/asterisk:1945 at 150.101.55.194/${EXTEN}@default) 

exten => 999,1,wait(1)
exten => 999,2,VoicemailMain(${CALLERIDNUM})
exten => 999,3,Hangup

[macro-std-exten]
exten => s,1,Dial(${ARG1},${ARG2})
exten => s,2,Voicemail(u${MACRO_EXTEN})
exten => s,3,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup






/etc/asterisk/modules.conf
==========================
[modules]
autoload=yes
noload => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
noload => app_intercom.so
load => chan_modem.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_skinny.so
load => chan_sccp.so
noload => chan_oh323.so

[global]
chan_modem.so=yes