[Asterisk-Users] Using * @ Home, all seems to work, but no sound to Softphone

Tue Mar 29 13:57:27 MST 2005

Hello,

To do some testing with Asterisk installed the latest Asterisk @ Home in a
Vmware system. All worked fine, I can access the web interface (AMP). I have
setup the extention and X-Lite softphone according to the description in the
Wike (http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite).

I can dial 200 (the softphone extention) and 1234 and they connect (the
softphone shows this, as well as the call record), but I don't get any sound
from it. I would expect to hear the Festival output that the asterisk
console shows it is generating.

I tried both X_lite and Firefly softphone, but phones do give me sounds when
pressing buttons etc, so it's not my loudspeaker ;-)

Any suggestions on what might be wrong?

Bas

Attached is the relevant output from the debug log:

Mar 29 02:46:11 WARNING[1433]: Inband DTMF is not supported on codec gsm.
Use RFC2833
Mar 29 02:46:11 DEBUG[1433]: Scheduling timer at 0 sample intervals
Mar 29 02:46:11 VERBOSE[1433]: == Spawn extension (macro-exten-vm, novm, 3)
exited non-zero on 'SIP/200-a5f0' in macro 'exten-vm'
Mar 29 02:46:11 VERBOSE[1433]: == Spawn extension (from-internal, 200, 1)
exited non-zero on 'SIP/200-a5f0'
Mar 29 02:46:11 VERBOSE[1433]: -- Executing
[1;36;40mMacro[0;37;40m("[1;35;40mSIP/200-a5f0[0;37;40m",
"[1;35;40mhangupcall[0;37;40m") in new stack
Mar 29 02:46:12 VERBOSE[1433]: -- Executing
[1;36;40mResetCDR[0;37;40m("[1;35;40mSIP/200-a5f0[0;37;40m",
"[1;35;40mw[0;37;40m") in new stack
Mar 29 02:46:12 DEBUG[1433]: cdr_mysql: inserting a CDR record.
Mar 29 02:46:12 DEBUG[1433]: cdr_mysql: SQL command as follows: INSERT INTO
cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
,billsec,disposition,amaflags,accountcode) VALUES ('2005-03-29
02:46:03','\"Bas Rijniersce\" <200>','200','200','from-internal',
'SIP/200-a5f0','','ResetCDR','w',9,8,'ANSWERED',3,'')
Mar 29 02:46:12 VERBOSE[1433]: -- Executing
[1;36;40mNoCDR[0;37;40m("[1;35;40mSIP/200-a5f0[0;37;40m",
"[1;35;40m[0;37;40m") in new stack
Mar 29 02:46:12 WARNING[1433]: CDR on channel 'SIP/200-a5f0' not posted
Mar 29 02:46:12 WARNING[1433]: CDR on channel 'SIP/200-a5f0' lacks end
Mar 29 02:46:12 VERBOSE[1433]: -- Executing
[1;36;40mWait[0;37;40m("[1;35;40mSIP/200-a5f0[0;37;40m",
"[1;35;40m5[0;37;40m") in new stack
Mar 29 02:46:12 VERBOSE[1433]: == Spawn extension (macro-hangupcall, s, 3)
exited non-zero on 'SIP/200-a5f0' in macro 'hangupcall'
Mar 29 02:46:12 VERBOSE[1433]: == Spawn extension (from-internal, h, 1)
exited non-zero on 'SIP/200-a5f0'
Mar 29 02:46:12 DEBUG[1433]: update_user_counter(200) - decrement inUse
counter
Mar 29 02:46:13 DEBUG[1433]: Auto destroying call
'276880640a18d769 at YnJpam5ib3g.'
Mar 29 02:46:15 DEBUG[1433]: Setting NAT on RTP to 0
Mar 29 02:46:15 DEBUG[1433]: Stopping retransmission on
'c14b2b4ab941be5e at YnJpam5ib3g.' of Response 1: Found
Mar 29 02:46:15 DEBUG[1433]: Setting NAT on RTP to 0
Mar 29 02:46:15 DEBUG[1433]: Check for res for 200
Mar 29 02:46:15 DEBUG[1433]: Call from user '200' is 1 out of 0
Mar 29 02:46:15 DEBUG[1433]: build_route: Contact hop:
Mar 29 02:46:15 VERBOSE[1433]: -- Executing
[1;36;40mAnswer[0;37;40m("[1;35;40mSIP/200-74a5[0;37;40m",
"[1;35;40m[0;37;40m") in new stack
Mar 29 02:46:15 VERBOSE[1433]: -- Executing
[1;36;40mAGI[0;37;40m("[1;35;40mSIP/200-74a5[0;37;40m",
"[1;35;40mfestival-script.pl|Welcome to the wonderful world of Asterisk!
Your phone number is 200.[0;37;40m") in new stack
Mar 29 02:46:15 VERBOSE[1433]: -- Launched AGI Script
/var/lib/asterisk/agi-bin/festival-script.pl

I also tried without forcing the gsm codec. Didn't make a difference