Festival with asterisk 1.6.1.11

mudcow007 · December 15, 2009, 4:09pm

i have installed festival onto my new install of Asterisk

i have got the festival server running

i have the following in my dial plan

exten => 1011,1,Answer exten => 1011,n,Festival(Hello Asterisk) exten => 1011,n,Hangup()

when i dial 1011 the asterisk cli says

Connected to Asterisk 1.6.1.11 currently running on pbx (pid = 3086) Verbosity is at least 10000 == Using SIP RTP CoS mark 5 -- Executing [1011@Operator:1] Answer("SIP/1010-0000000b", "") in new stack -- Executing [1011@Operator:2] Festival("SIP/1010-0000000b", "Hello Asterisk") in new stack == Parsing '/etc/asterisk/festival.conf': == Found pbx-2*CLI>

the festival server says root@pbx:~# festival --server server Tue Dec 15 16:09:02 2009 : Festival server started on port 1314 client(1) Tue Dec 15 16:09:05 2009 : accepted from pbx- client(1) Tue Dec 15 16:09:05 2009 : disconnected

although on the handset i hear no speech at all and the call doesnt end?
do you think it could be a codec issue??

mudcow007 · December 16, 2009, 8:59am

anyone?

mudcow007 · December 16, 2009, 11:06am

hello all im adding a sip debug i did during a test call

hopefully it shoudl shed soem light on the issue with someone

[code]SIP Debugging enabled
pbx*CLI>
<— SIP read from UDP://192.168.1.91:5060 —>
INVITE sip:1011@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK-67ab790b
From: “test2” sip:1001@192.168.1.20;tag=402e9e3752d9842bo0
To: sip:1011@192.168.1.20
Call-ID: 2beb4027-55bbcafb@192.168.1.91
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “test2” sip:1001@192.168.1.91:5060
Expires: 240
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 307831 307831 IN IP4 192.168.1.91
s=-
c=IN IP4 192.168.1.91
t=0 0
m=audio 16466 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (14 headers 11 lines) —
== Using SIP RTP CoS mark 5
Sending to 192.168.1.91 : 5060 (no NAT)
Using INVITE request as basis request - 2beb4027-55bbcafb@192.168.1.91
Found peer ‘1001’ for ‘1001’ from 192.168.1.91:5060

<— Reliably Transmitting (no NAT) to 192.168.1.91:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK-67ab790b;received=192.168.1.91
From: “test2” sip:1001@192.168.1.20;tag=402e9e3752d9842bo0
To: sip:1011@192.168.1.20;tag=as1b95db18
Call-ID: 2beb4027-55bbcafb@192.168.1.91
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6f332629"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘2beb4027-55bbcafb@192.168.1.91’ in 6400 ms (Method: INVITE)
pbx*CLI>
<— SIP read from UDP://192.168.1.91:5060 —>
ACK sip:1011@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK-67ab790b
From: “test2” sip:1001@192.168.1.20;tag=402e9e3752d9842bo0
To: sip:1011@192.168.1.20;tag=as1b95db18
Call-ID: 2beb4027-55bbcafb@192.168.1.91
CSeq: 101 ACK
Max-Forwards: 70
Contact: “test2” sip:1001@192.168.1.91:5060
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 0

<------------->
— (10 headers 0 lines) —
pbx*CLI>
<— SIP read from UDP://192.168.1.91:5060 —>
INVITE sip:1011@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK-f01f3497
From: “test2” sip:1001@192.168.1.20;tag=402e9e3752d9842bo0
To: sip:1011@192.168.1.20
Call-ID: 2beb4027-55bbcafb@192.168.1.91
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“6f332629”,uri="sip:1011@192.168.1.20",algorithm=MD5,response="b2a897cd8f3aa2f0502ccdd8e5f02343"
Contact: “test2” sip:1001@192.168.1.91:5060
Expires: 240
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 307831 307831 IN IP4 192.168.1.91
s=-
c=IN IP4 192.168.1.91
t=0 0
m=audio 16466 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (15 headers 11 lines) —
Sending to 192.168.1.91 : 5060 (no NAT)
Using INVITE request as basis request - 2beb4027-55bbcafb@192.168.1.91
Found peer ‘1001’ for ‘1001’ from 192.168.1.91:5060
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.91:16466
Looking for 1011 in Operator (domain 192.168.1.20)
list_route: hop: sip:1001@192.168.1.91:5060
pbx*CLI>
<— Transmitting (no NAT) to 192.168.1.91:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK-f01f3497;received=192.168.1.91
From: “test2” sip:1001@192.168.1.20;tag=402e9e3752d9842bo0
To: sip:1011@192.168.1.20
Call-ID: 2beb4027-55bbcafb@192.168.1.91
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:1011@192.168.1.20
Content-Length: 0

<------------>
– Executing [1011@Operator:1] Answer(“SIP/1001-00000016”, “”) in new stack
Audio is at 192.168.1.20 port 17046
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
pbx*CLI>
<— Reliably Transmitting (no NAT) to 192.168.1.91:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK-f01f3497;received=192.168.1.91
From: “test2” sip:1001@192.168.1.20;tag=402e9e3752d9842bo0
To: sip:1011@192.168.1.20;tag=as120b3f6a
Call-ID: 2beb4027-55bbcafb@192.168.1.91
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:1011@192.168.1.20
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 747902357 747902357 IN IP4 192.168.1.20
s=Asterisk PBX 1.6.1.11
c=IN IP4 192.168.1.20
t=0 0
m=audio 17046 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
pbx*CLI>
<— SIP read from UDP://192.168.1.91:5060 —>
ACK sip:1011@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK-cf0ae14f
From: “test2” sip:1001@192.168.1.20;tag=402e9e3752d9842bo0
To: sip:1011@192.168.1.20;tag=as120b3f6a
Call-ID: 2beb4027-55bbcafb@192.168.1.91
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“6f332629”,uri="sip:1011@192.168.1.20",algorithm=MD5,response="b2a897cd8f3aa2f0502ccdd8e5f02343"
Contact: “test2” sip:1001@192.168.1.91:5060
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 0

<------------->
— (11 headers 0 lines) —
– Executing [1011@Operator:2] Festival(“SIP/1001-00000016”, “Hello world”) in new stack
== Parsing ‘/etc/asterisk/festival.conf’: == Found
pbx*CLI>
<— SIP read from UDP://192.168.1.91:5060 —>
BYE sip:1011@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK-f2885f20
From: “test2” sip:1001@192.168.1.20;tag=402e9e3752d9842bo0
To: sip:1011@192.168.1.20;tag=as120b3f6a
Call-ID: 2beb4027-55bbcafb@192.168.1.91
CSeq: 103 BYE
Max-Forwards: 70
Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“6f332629”,uri="sip:1011@192.168.1.20",algorithm=MD5,response="c24dacc239386a093b4dc00ad9bccfde"
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.1.91 : 5060 (no NAT)
pbx*CLI>
<— Transmitting (no NAT) to 192.168.1.91:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.91:5060;branch=z9hG4bK-f2885f20;received=192.168.1.91
From: “test2” sip:1001@192.168.1.20;tag=402e9e3752d9842bo0
To: sip:1011@192.168.1.20;tag=as120b3f6a
Call-ID: 2beb4027-55bbcafb@192.168.1.91
CSeq: 103 BYE
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>

[/code]

system · April 14, 2011, 11:14am

Hi! Same issue here. Did you find a solution?

ketan.jadhav · April 14, 2011, 1:34pm

Hi,

Can you please try the text2wav from the terminal and check whether text2wav is installed properly or not