No voice for first 15 secs

Asterisk Asterisk Support
1

hiii
i can connect the call but caller unable to listen the voice first 15 sec
here is the my sip debug log .
does anyone have any idea…

Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK10101;received=192.168.1.102
From: 502181074608074 sip:502181074608074@192.168.1.102;tag=iqinl
To: sip:234159262856399@192.168.1.102;tag=as40769ba0
Call-ID: 1612824113@192.168.1.102
CSeq: 600 INVITE
Server: Asterisk PBX 1.6.2.7-1ubuntu1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:234159262856399@192.168.1.102
Content-Type: application/sdp
Content-Length: 219

v=0
o=root 1018073526 1018073526 IN IP4 192.168.1.102
s=Asterisk PBX 1.6.2.7-1ubuntu1.2
c=IN IP4 192.168.1.102
t=0 0
m=audio 18964 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Playing ‘prepaid-you-have’ (escape_digits=#) (sample_offset 0)
Retransmitting #1 (no NAT) to 192.168.1.102:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK10101;received=192.168.1.102
From: 502181074608074 sip:502181074608074@192.168.1.102;tag=iqinl
To: sip:234159262856399@192.168.1.102;tag=as40769ba0
Call-ID: 1612824113@192.168.1.102
CSeq: 600 INVITE
Server: Asterisk PBX 1.6.2.7-1ubuntu1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:234159262856399@192.168.1.102
Content-Type: application/sdp
Content-Length: 219

v=0
o=root 1018073526 1018073526 IN IP4 192.168.1.102
s=Asterisk PBX 1.6.2.7-1ubuntu1.2
c=IN IP4 192.168.1.102
t=0 0
m=audio 18964 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- <SIP/234159262856399-0000005f> Playing 'digits/20.gsm' (language 'en')

openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
ACK sip:234159262856399@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK34154
From: 502181074608074 sip:502181074608074@192.168.1.102;tag=iqinl
To: sip:234159262856399@192.168.1.102;tag=
Call-ID: 1612824113@192.168.1.102
CSeq: 600 INVITE
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
– Playing ‘dollars’ (escape_digits=#) (sample_offset 0)
– Playing ‘vm-and’ (escape_digits=#) (sample_offset 0)
Retransmitting #6 (no NAT) to 192.168.1.102:5062:
CANCEL sip:234159262856399@192.168.1.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK16391e9a;rport
Max-Forwards: 70
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as516b5284
To: sip:234159262856399@192.168.1.102:5062
Call-ID: 6aee20092a87b503064cca753d2c0c4d@192.168.1.102
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1.2
Content-Length: 0

-- <SIP/234159262856399-0000005f> Playing 'digits/80.gsm' (language 'en')
-- <SIP/234159262856399-0000005f> Playing 'digits/2.gsm' (language 'en')
-- Playing 'prepaid-cents' (escape_digits=#) (sample_offset 0)
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/234159262856399-0000005f> Playing 'digits/9.gsm' (language 'en')

[Oct 4 13:51:31] WARNING[2310]: chan_sip.c:3906 retrans_pkt: Maximum retries exceeded on transmission 6aee20092a87b503064cca753d2c0c4d@192.168.1.102 for seqno 102 (Non-critical Request) – See doc/sip-retransmit.txt.
– <SIP/234159262856399-0000005f> Playing ‘digits/hundred.gsm’ (language ‘en’)
– <SIP/234159262856399-0000005f> Playing ‘digits/5.gsm’ (language ‘en’)
– Playing ‘prepaid-minutes’ (escape_digits=#) (sample_offset 0)
– AGI Script Executing Application: (DIAL) Options: (SIP/234159262856399,60,HRrL(36000000:61000:30000))
– Limit Data for this call:
> timelimit = 36000000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP RTP CoS mark 5
Audio is at 192.168.1.102 port 14334
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 192.168.1.102:5062:
INVITE sip:234159262856399@192.168.1.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3b915302;rport
Max-Forwards: 70
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as07ffc870
To: sip:234159262856399@192.168.1.102:5062
Contact: sip:502181074608074@192.168.1.102
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1.2
Date: Tue, 04 Oct 2011 12:51:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 256315263 256315263 IN IP4 192.168.1.102
s=Asterisk PBX 1.6.2.7-1ubuntu1.2
c=IN IP4 192.168.1.102
t=0 0
m=audio 14334 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called 234159262856399

Retransmitting #1 (no NAT) to 192.168.1.102:5062:
INVITE sip:234159262856399@192.168.1.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3b915302;rport
Max-Forwards: 70
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as07ffc870
To: sip:234159262856399@192.168.1.102:5062
Contact: sip:502181074608074@192.168.1.102
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1.2
Date: Tue, 04 Oct 2011 12:51:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 256315263 256315263 IN IP4 192.168.1.102
s=Asterisk PBX 1.6.2.7-1ubuntu1.2
c=IN IP4 192.168.1.102
t=0 0
m=audio 14334 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK16391e9a;rport
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as516b5284
To: sip:234159262856399@192.168.1.102:5062
Call-ID: 6aee20092a87b503064cca753d2c0c4d@192.168.1.102
CSeq: 102 INVITE
Contact: sip:234159262856399@192.168.1.102
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Retransmitting #2 (no NAT) to 192.168.1.102:5062:
INVITE sip:234159262856399@192.168.1.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3b915302;rport
Max-Forwards: 70
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as07ffc870
To: sip:234159262856399@192.168.1.102:5062
Contact: sip:502181074608074@192.168.1.102
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1.2
Date: Tue, 04 Oct 2011 12:51:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 256315263 256315263 IN IP4 192.168.1.102
s=Asterisk PBX 1.6.2.7-1ubuntu1.2
c=IN IP4 192.168.1.102
t=0 0
m=audio 14334 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
BYE sip:502181074608074@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK40079
From: 234159262856399 sip:234159262856399@192.168.1.102;tag=as516b5284
To: 234159262856399 sip:502181074608074@192.168.1.102;tag=egnhk
Call-ID: 6aee20092a87b503064cca753d2c0c4d@192.168.1.102
CSeq: 58 BYE
Contact: 234159262856399 sip:234159262856399@192.168.1.102:5062
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.102 : 5062 (no NAT)
Scheduling destruction of SIP dialog ‘6aee20092a87b503064cca753d2c0c4d@192.168.1.102’ in 32000 ms (Method: BYE)

<— Transmitting (no NAT) to 192.168.1.102:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK40079;received=192.168.1.102
From: 234159262856399 sip:234159262856399@192.168.1.102;tag=as516b5284
To: 234159262856399 sip:502181074608074@192.168.1.102;tag=egnhk
Call-ID: 6aee20092a87b503064cca753d2c0c4d@192.168.1.102
CSeq: 58 BYE>
Server: Asterisk PBX 1.6.2.7-1ubuntu1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
BYE sip:502181074608074@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK13041
From: 234159262856399 sip:234159262856399@192.168.1.102;tag=as516b5284
To: 234159262856399 sip:502181074608074@192.168.1.102;tag=
Call-ID: 6aee20092a87b503064cca753d2c0c4d@192.168.1.102
CSeq: 58 BYE
Contact: 234159262856399 sip:234159262856399@192.168.1.102:5062
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.102 : 5062 (no NAT)
Scheduling destruction of SIP dialog ‘6aee20092a87b503064cca753d2c0c4d@192.168.1.102’ in 32000 ms (Method: BYE)

<— Transmitting (no NAT) to 192.168.1.102:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK13041;received=192.168.1.102
From: 234159262856399 sip:234159262856399@192.168.1.102;tag=as516b5284
To: 234159262856399 sip:502181074608074@192.168.1.102;tag=
Call-ID: 6aee20092a87b503064cca753d2c0c4d@192.168.1.102
CSeq: 58 BYE
Server: Asterisk PBX 1.6.2.7-1ubuntu1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
Retransmitting #3 (no NAT) to 192.168.1.102:5062:
INVITE sip:234159262856399@192.168.1.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3b915302;rport
Max-Forwards: 70
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as07ffc870
To: sip:234159262856399@192.168.1.102:5062
Contact: sip:502181074608074@192.168.1.102
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1.2
Date: Tue, 04 Oct 2011 12:51:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 256315263 256315263 IN IP4 192.168.1.102
s=Asterisk PBX 1.6.2.7-1ubuntu1.2
c=IN IP4 192.168.1.102
t=0 0
m=audio 14334 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK16391e9a;rport
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as516b5284
To: sip:234159262856399@192.168.1.102:5062
Call-ID: 6aee20092a87b503064cca753d2c0c4d@192.168.1.102
CSeq: 102 INVITE
Contact: sip:234159262856399@192.168.1.102
Content-Length: 0

<------------->
— (8 headers 0 lines) —
openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3b915302;rport
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as07ffc870
To: sip:234159262856399@192.168.1.102:5062
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 102 INVITE
Contact: sip:234159262856399@192.168.1.102
Content-Length: 0

<------------->
— (8 headers 0 lines) —
openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
BYE sip:502181074608074@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK95200
From: 234159262856399 sip:234159262856399@192.168.1.102;tag=as516b5284
To: 234159262856399 sip:502181074608074@192.168.1.102;tag=
Call-ID: 6aee20092a87b503064cca753d2c0c4d@192.168.1.102
CSeq: 58 BYE
Contact: 234159262856399 sip:234159262856399@192.168.1.102:5062
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.102 : 5062 (no NAT)
Scheduling destruction of SIP dialog ‘6aee20092a87b503064cca753d2c0c4d@192.168.1.102’ in 32000 ms (Method: BYE)

<— Transmitting (no NAT) to 192.168.1.102:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK95200;received=192.168.1.102
From: 234159262856399 sip:234159262856399@192.168.1.102;tag=as516b5284
To: 234159262856399 sip:502181074608074@192.168.1.102;tag=
Call-ID: 6aee20092a87b503064cca753d2c0c4d@192.168.1.102
CSeq: 58 BYE
Server: Asterisk PBX 1.6.2.7-1ubuntu1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3b915302;rport
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as07ffc870
To: sip:234159262856399@192.168.1.102:5062;tag=rcmrc
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 102 INVITE
Contact: sip:234159262856399@192.168.1.102
Content-Length: 0

<------------->
— (8 headers 0 lines) —
– SIP/234159262856399-00000060 is ringing
openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3b915302;rport
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as07ffc870
To: sip:234159262856399@192.168.1.102:5062;tag=ewpol
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 102 INVITE
Contact: sip:234159262856399@192.168.1.102
Content-Length: 0

<------------->
— (8 headers 0 lines) —
– SIP/234159262856399-00000060 is ringing
openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3b915302;rport
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as07ffc870
To: sip:234159262856399@192.168.1.102:5062;tag=ewpol
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 102 INVITE
Contact: sip:234159262856399@192.168.1.102:5062;expires=3600
Content-Type: application/sdp
Content-Length: 139

v=0
o=234159262856399 0 0 IN IP4 192.168.1.102
s=Talk Time
t=0 0
m=audio 16504 RTP/AVP 3
c=IN IP4 192.168.1.102
a=rtpmap:3 GSM/8000

<------------->
— (9 headers 7 lines) —
Found RTP audio format 3
Found audio description format GSM for ID 3
Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.102:16504
list_route: hop: sip:234159262856399@192.168.1.102:5062
set_destination: Parsing sip:234159262856399@192.168.1.102:5062 for address/port to send to
set_destination: set destination to 192.168.1.102, port 5062
Transmitting (no NAT) to 192.168.1.102:5062:
ACK sip:234159262856399@192.168.1.102:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK60397a1c;rport
Max-Forwards: 70
From: “502181074608074” sip:502181074608074@192.168.1.102;tag=as07ffc870
To: sip:234159262856399@192.168.1.102:5062;tag=ewpol
Contact: sip:502181074608074@192.168.1.102
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1.2
Content-Length: 0

-- SIP/234159262856399-00000060 answered SIP/234159262856399-0000005f
-- Packet2Packet bridging SIP/234159262856399-0000005f and SIP/234159262856399-00000060

Really destroying SIP dialog ‘6aee20092a87b503064cca753d2c0c4d@192.168.1.102’ Method: BYE
openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
BYE sip:502181074608074@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK72189
From: 234159262856399 sip:234159262856399@192.168.1.102;tag=as07ffc870
To: 234159262856399 sip:502181074608074@192.168.1.102;tag=ewpol
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 842 BYE
Contact: 234159262856399 sip:234159262856399@192.168.1.102:5062
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.102 : 5062 (no NAT)
openbts1*CLI>
<— Transmitting (no NAT) to 192.168.1.102:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK72189;received=192.168.1.102
From: 234159262856399 sip:234159262856399@192.168.1.102;tag=as07ffc870
To: 234159262856399 sip:502181074608074@192.168.1.102;tag=ewpol
Call-ID: 76c5c0da66965e576e46a1a76209dcf9@192.168.1.102
CSeq: 842 BYE
Server: Asterisk PBX 1.6.2.7-1ubuntu1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
– <SIP/234159262856399-0000005f> Playing ‘prepaid-enter-dest.gsm’ (language ‘en’)
Really destroying SIP dialog ‘76c5c0da66965e576e46a1a76209dcf9@192.168.1.102’ Method: BYE
openbts1*CLI>
<— SIP read from UDP:192.168.1.102:5062 —>
BYE sip:234159262856399@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK48747
From: 502181074608074 sip:502181074608074@192.168.1.102;tag=iqinl
To: 502181074608074 sip:234159262856399@192.168.1.102;tag=
Call-ID: 1612824113@192.168.1.102
CSeq: 601 BYE
Contact: 502181074608074 sip:502181074608074@192.168.1.102:5062
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.102 : 5062 (no NAT)
openbts1*CLI>
<— Transmitting (no NAT) to 192.168.1.102:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK48747;received=192.168.1.102
From: 502181074608074 sip:502181074608074@192.168.1.102;tag=iqinl
To: 502181074608074 sip:234159262856399@192.168.1.102;tag=
Call-ID: 1612824113@192.168.1.102
CSeq: 601 BYE
Server: Asterisk PBX 1.6.2.7-1ubuntu1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
– <SIP/234159262856399-0000005f>AGI Script a2billing.php completed, returning -1
Really destroying SIP dialog ‘1612824113@192.168.1.102’ Method: BYE

2

You have three different conversations that are having retransmitted frames (one on the sixth attempt). I would try and fix that first. The obvious reason would be that your network is overloaded.

3

thanks for reply
i am using the two mobiles connected , your saying that network overload. ??

4

You are losing a lot of packets. Either the network is overloaded, or you have some systematic misconfiguration that causes all of certain types of packet to get lost. The latter would normally result in the call dropping after about 30 seconds, so I suspect the former.

In any case, you need to find out why you have so many retransmissions.

5

A second good reason would be that the SIP phones are not working as they should … (network is hardly ever a problem, usually it is a SIP phone issue).