THanks for your reply
i am using a eyebeam soft phone
and yes i am getting a incoming ring on my soft phone but the file which plays after that is DEMO-congrats from asterisk default sounds
i am also placing my my sip logs in which i have highlighted that part
also i am using ztdummy which i belive is working i checked it thru lsmod command i am currently trying to test it on on 2 seats after that i will exapnd it
please lemme know if you need any other info from my end
thanks again for your help
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.3.15:6182
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex |ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 ( nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event) , combined - 0x1 (telephone-event)
list_route: hop: <sip:3003@203.122.26.234:27076>
set_destination: Parsing <sip:3003@203.122.26.234:27076> for address/port to sen d to
set_destination: set destination to 203.122.26.234, port 27076
Transmitting (NAT) to 203.122.26.234:27076:
ACK sip:3003@203.122.26.234:27076 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK0570e8b9;rport
From: "S0609152242098600051" <sip:asterisk@203.122.26.232>;tag=as65645334
To: <sip:3003@203.122.26.234:27076>;tag=4e04391d
Contact: <sip:asterisk@203.122.26.232>
Call-ID:
11b82fbf708ced7a24ed12310d998b92@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
> Channel SIP/3003-084b7338 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
== Starting SIP/3003-084b7338 at default,8600051,1 failed so falling back to e xten 's'
-- Executing Wait("SIP/3003-084b7338", "1") in new stack
-- Executing Answer("SIP/3003-084b7338", "") in new stack
-- Executing Set("SIP/3003-084b7338", "TIMEOUT(digit)=5") in new stack
-- Digit timeout set to 5
-- Executing Set("SIP/3003-084b7338", "TIMEOUT(response)=10") in new stack
-- Response timeout set to 10
--
*********************************************************************************************************************************
Executing BackGround("SIP/3003-084b7338", "demo-congrats") in new stack
*************************************************************************************************************************************
-- Playing 'demo-congrats' (language 'en')
sip*CLI> sip d
<-- SIP read from 203.122.26.234:27076:
--- (0 headers 0 lines) Nat keepalive ---
sip*CLI> sip dbeug
No such command 'sip dbeug' (type 'help' for help)
sip*CLI>
<-- SIP read from 203.122.26.234:4148:
--- (0 headers 0 lines) Nat keepalive ---
sip*CLI>
<-- SIP read from 203.122.26.234:27076:
BYE sip:asterisk@203.122.26.232:5060 SIP/2.0
To: "S0609152242098600051"<sip:asterisk@203.122.26.232>;tag=as65645334
From: <sip:3003@203.122.26.234:27076>;tag=4e04391d
Via: SIP/2.0/UDP 203.122.26.234:27076;branch=z9hG4bK-d87543-214828442-1--d87543-;rport
Call-ID:
11b82fbf708ced7a24ed12310d998b92@203.122.26.232
CSeq: 2 BYE
Contact: <sip:3003@203.122.26.234:27076>
Max-Forwards: 70
User-Agent: eyeBeam release 3007n stamp 17816
Content-Length: 0
--- (10 headers 0 lines)---
Sending to 203.122.26.234 : 27076 (NAT)
Transmitting (NAT) to 203.122.26.234:27076:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.122.26.234:27076;branch=z9hG4bK-d87543-214828442-1--d87543-;received=203.122.26.234;rport=27076
From: <sip:3003@203.122.26.234:27076>;tag=4e04391d
To: "S0609152242098600051"<sip:asterisk@203.122.26.232>;tag=as65645334
Call-ID:
11b82fbf708ced7a24ed12310d998b92@203.122.26.232
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:asterisk@203.122.26.232>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
== Spawn extension (default, s, 5) exited non-zero on 'SIP/3003-084b7338'
Destroying call
'11b82fbf708ced7a24ed12310d998b92@203.122.26.232'
sip*CLI>
<-- SIP read from 203.122.26.234:27076:
--- (0 headers 0 lines) Nat keepalive ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
We're at 203.122.26.232 port 19880
Adding codec 0x40 (slin) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 21 lines
Reliably Transmitting (NAT) to 203.122.26.234:27076:
INVITE sip:3003@203.122.26.234:27076 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK4e94039c;rport
From: "ACagcW11583403496666" <sip:asterisk@203.122.26.232>;tag=as61d01c4f
To: <sip:3003@203.122.26.234:27076>
Contact: <sip:asterisk@203.122.26.232>
Call-ID:
1623fe4057d53b8529cc80556fc30d20@203.122.26.232
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Sep 2006 17:12:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 494
v=0
o=root 2296 2296 IN IP4 203.122.26.232
s=session
c=IN IP4 203.122.26.232
t=0 0
m=audio 19880 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101
a=rtpmap:10 L16/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -