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Hi Folks,
I have asterisk 1.4.18 on REMOTE suse Linux enterprise 10. Also this machine has Digium TDM2400P card installed.
I'm using X-Lite here as a soft phone on windows machine which is under different LAN. My X-LITE registered to remote machine via it's Public IP.
Now my Problem is,
I have installed festival 1.95 via rpm on remote suse 10.
after that I written following code in extensions.conf,
exten => 123,1,Answer()
exten => 123,2,Festival("Hi How are you")
exten => 123,3,Hangup()
asterisk shows me following things on console,
-- Executing [9009@sip:1] Answer("SIP/101-081ea6e8", "") in new stack
-- Executing [9009@sip:2] Festival("SIP/101-081ea6e8", "hi How are you") in new stack
== Parsing '/etc/asterisk/festival.conf': Found
== Spawn extension (sip, 9009, 2) exited non-zero on 'SIP/101-081ea6e8'
[Mar 19 10:48:42] WARNING[19352]: chan_sip.c:1947 retrans_pkt: Maximum retries exceeded on transmission Y2QxMmY0ODkxZTdmYjY3YTQ3Yjg5NTU3YTljZWZlMzk. for seqno 2 (Critical Response)
RTP-stats
* Our Receiver:
SSRC: 0
Received packets: 0
Lost packets: 0
Jitter: 0.0000
Transit: 0.0000
RR-count: 0
* Our Sender:
SSRC: 1443552997
Sent packets: 0
Lost packets: 0
Jitter: 0
SR-count: 0
RTT: 0.000000
My Festival showed me following log,
festival --server
server Wed Mar 19 10:47:29 2008 : Festival server started on port 1314
client(1) Wed Mar 19 10:48:22 2008 : accepted from localhost
client(1) Wed Mar 19 10:48:22 2008 : disconnected
Note: Festival gets disconnected immediately.
my sip.conf is as follows, I've registered x-lite to 101.
[101]
callerid=101 <101>
canreinvite=no
dtmfmode=rfc2833
host=dynamic
nat=yes
port=5060
bindport=5060
bindaddr=0.0.0.0
qualify=yes
record_in=Adhoc
record_out=Adhoc
secret=101
type=friend
context=sip
username=101
allow=all
[As my Suse machine is on remote ip I've enabled NAT and Qualify under this configuration]
Problem here is :
I'm not able to hear sound "Hi how are you" on x-lite.
when it comes to festival asterisk is directly gives following error,
Spawn extension (sip, 9009, 2) exited non-zero on 'SIP/101-081ea6e8'
and it does not go to 3rd priority to hangup.
other than this I was also receiving same problem with Cepstral TTS also. I formatted machine and now just trying with Festival.
My second problem is,
When I'm also using PHPAGI for writing agis,
Now if I use text2wav function from php then it works fine, As it creates file first and then plays it.
But when I specify escape digits to text2wav, and pressed one while its playing message. asterisk does not interpret that.
Eg:
if I write following code,
$agi = new AGI();
$agi->text2wav("Enter your choice here.","0 1 #");
and while it plays "Enter your choice here." if we enter 0 or 1 or # then it should stop playing and return digit pressed.
Its not doing so.
This problem I'm facing for remote machine only if I create same environment here then asterisk works fine on that.
also if we put the machine with same environment on Public IP then also it's working fine.
But only Remote machine has problems.
Any help regarding this will be really helpful,
So please help me out in this.
Thanks in advance,
Regards,
Rahul |