Asterisk Disconnect Call around 90 second

Asterisk Asterisk Support
1

Dear all,

I just install asterisk 1.8.8.1 with A2billing. when I am calling to mobile number, it always disconnect a call around 90s during talking. I don’t know what is wrong with asterisk. I think it should be any wrong configure or miss configure. please help. Below is debug mode in asterisk

<— SIP read from UDP:xxx.xxx.xxx.x:5060 —>
BYE sip:179899101234@192.168.0.73:5060 SIP/2.0
Call-ID: 5ae058df31f0c9666f12394e3076d85c@192.168.0.73:5060
CSeq: 104 BYE
From: sip:85510608707@xxx.xxx.xxx.x;tag=2E012328-903
To: “2934398838” sip:179899101234@192.168.0.73;tag=as0d34a3ee
Via: SIP/2.0/UDP xxx.xxx.xxx.x:5060;branch=z9hG4bK1326958876428
Max-Forwards: 50
Reason: SIP;cause=200;text="Call Snapped"
Content-Length: 0

Regards,
Sokphak

2

That trace is rather short, but seems to indicate the problem is not in Asterisk, although I don’t really know what the unidentified remote side means by “call snapped”.

3

Yes, it always disconnect call during talking every call. Always less than 2mn and after that if I try to call again. It should me “Everyone is busy/congested” I need to reload in CLI mode to start call again. Someone know any tool/comment/debug to find the exactly problem? Thanks

4

You need the documentation and logs for the remote SIP system. Asterisk is not disconnecting, it is the remote system that is initiating the disconnect.

I don’t know why restarting Asterisk is needed to resolve the lockout, but the evidence provided so far does not show Asterisk doing anything wrong.

More generally, if you are providing SIP traces, you need to provide the trace for the whole call.

5

Hello,

Below is my debug log in asterisk for one call. It always disconnect call. Please help to check and advise.

<— SIP read from UDP:192.168.1.66:13438 —>
INVITE sip:85510608707@192.168.0.68 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-87618032ef04a207-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:6640757506@192.168.1.66:13438
To: "85510608707"sip:85510608707@192.168.0.68
From: "6640757506"sip:6640757506@192.168.0.68;tag=e61f5e13
Call-ID: YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 429

v=0
o=- 6 2 IN IP4 192.168.1.66
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.66
t=0 0
m=audio 49756 RTP/AVP 18 101
a=alt:1 4 : 9MjV0Vd8 9bg1KcuS 192.168.1.66 49756
a=alt:2 3 : 4kx/NrP5 5z04iCCW 10.1.1.25 49756
a=alt:3 2 : uH6JgL4X AWtrqQiN 192.168.168.1 49756
a=alt:4 1 : Yvv2uowi Gb1Ds2o7 192.168.157.1 49756
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (12 headers 15 lines) —
Sending to 192.168.1.66:13438 (NAT)
Using INVITE request as basis request - YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.
Found peer ‘6640757506’ for ‘6640757506’ from 192.168.1.66:13438

<— Reliably Transmitting (NAT) to 192.168.1.66:13438 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-87618032ef04a207-1—d8754z-;received=192.168.1.66;rport=13438
From: "6640757506"sip:6640757506@192.168.0.68;tag=e61f5e13
To: "85510608707"sip:85510608707@192.168.0.68;tag=as5016dc32
Call-ID: YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3e4dc1e1"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.66:13438 —>
ACK sip:85510608707@192.168.0.68 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-87618032ef04a207-1—d8754z-;rport
To: "85510608707"sip:85510608707@192.168.0.68;tag=as5016dc32
From: "6640757506"sip:6640757506@192.168.0.68;tag=e61f5e13
Call-ID: YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.
CSeq: 1 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:192.168.1.66:13438 —>
INVITE sip:85510608707@192.168.0.68 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-f84ea6250b0e840f-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:6640757506@192.168.1.66:13438
To: "85510608707"sip:85510608707@192.168.0.68
From: “6640757506"sip:6640757506@192.168.0.68;tag=e61f5e13
Call-ID: YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“6640757506”,realm=“asterisk”,nonce=“3e4dc1e1”,uri="sip:85510608707@192.168.0.68”,response=“cfb177ccff4ec1883a82357c34f28f6e”,algorithm=MD5
Content-Length: 429

v=0
o=- 6 2 IN IP4 192.168.1.66
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.66
t=0 0
m=audio 49756 RTP/AVP 18 101
a=alt:1 4 : 9MjV0Vd8 9bg1KcuS 192.168.1.66 49756
a=alt:2 3 : 4kx/NrP5 5z04iCCW 10.1.1.25 49756
a=alt:3 2 : uH6JgL4X AWtrqQiN 192.168.168.1 49756
a=alt:4 1 : Yvv2uowi Gb1Ds2o7 192.168.157.1 49756
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (13 headers 15 lines) —
Sending to 192.168.1.66:13438 (NAT)
Using INVITE request as basis request - YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.
Found peer ‘6640757506’ for ‘6640757506’ from 192.168.1.66:13438
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.66:49756
Looking for 85510608707 in a2billing (domain 192.168.0.68)
list_route: hop: sip:6640757506@192.168.1.66:13438

<— Transmitting (NAT) to 192.168.1.66:13438 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-f84ea6250b0e840f-1—d8754z-;received=192.168.1.66;rport=13438
From: "6640757506"sip:6640757506@192.168.0.68;tag=e61f5e13
To: "85510608707"sip:85510608707@192.168.0.68
Call-ID: YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:85510608707@192.168.0.68:5060
Content-Length: 0

<------------>
– Executing [85510608707@a2billing:1] NoOp(“SIP/6640757506-0000002c”, “A2Billing Start”) in new stack
– Executing [85510608707@a2billing:2] AGI(“SIP/6640757506-0000002c”, “a2billing.php,1”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
– AGI Script Executing Application: (DIAL) Options: (SIP/DATA4ICT/85510608707,60,HRL(1514000:61000:30000))
> Limit Data for this call:
> timelimit = 1514000 ms (1514.000 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP RTP CoS mark 5
Audio is at 5060
Video is at 192.168.0.68:5060
Adding codec 0x100 (g729) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding video codec 0x100000 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to xxx.xxx.xxx.xx:5060:
INVITE sip:85510608707@xxx.xxx.xxx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK7a9abfd9;rport
Max-Forwards: 70
From: “6640757506” sip:179899101234@192.168.0.68;tag=as65768d12
To: sip:85510608707@xxx.xxx.xxx.xx
Contact: sip:179899101234@192.168.0.68:5060
Call-ID: 6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.8.1
Date: Sat, 21 Jan 2012 00:59:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 390

v=0
o=root 2027046490 2027046490 IN IP4 192.168.0.68
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.68
b=CT:384
t=0 0
m=audio 18838 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10908 RTP/AVP 34 98
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=sendrecv

-- Called SIP/DATA4ICT/85510608707

<— SIP read from UDP:xxx.xxx.xxx.xx:5060 —>
SIP/2.0 407 Proxy Authentication required
Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK7a9abfd9;rport
From: “6640757506” sip:179899101234@192.168.0.68;tag=as65768d12
To: sip:85510608707@xxx.xxx.xxx.xx
Call-ID: 6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“xxx.xxx.xxx.xx”,nonce=“c091913e78a9b9a20fb6c044e1db6b10”,opaque="",stale=FALSE,algorithm=MD5
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to xxx.xxx.xxx.xx:5060:
ACK sip:85510608707@xxx.xxx.xxx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK7a9abfd9;rport
Max-Forwards: 70
From: “6640757506” sip:179899101234@192.168.0.68;tag=as65768d12
To: sip:85510608707@xxx.xxx.xxx.xx
Contact: sip:179899101234@192.168.0.68:5060
Call-ID: 6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.8.1
Content-Length: 0

Audio is at 5060
Video is at 192.168.0.68:5060
Adding codec 0x100 (g729) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding video codec 0x100000 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to xxx.xxx.xxx.xx:5060:
INVITE sip:85510608707@xxx.xxx.xxx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK621ca390;rport
Max-Forwards: 70
From: “6640757506” sip:179899101234@192.168.0.68;tag=as65768d12
To: sip:85510608707@xxx.xxx.xxx.xx
Contact: sip:179899101234@192.168.0.68:5060
Call-ID: 6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.8.1
Proxy-Authorization: Digest username=“179899101234”, realm=“xxx.xxx.xxx.xx”, algorithm=MD5, uri=“sip:85510608707@xxx.xxx.xxx.xx”, nonce=“c091913e78a9b9a20fb6c044e1db6b10”, response="d3aface9c19700d867b77635838c29e7"
Date: Sat, 21 Jan 2012 00:59:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 390

v=0
o=root 2027046490 2027046491 IN IP4 192.168.0.68
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.68
b=CT:384
t=0 0
m=audio 18838 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10908 RTP/AVP 34 98
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=sendrecv

<— SIP read from UDP:xxx.xxx.xxx.xx:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK621ca390;rport
From: “6640757506” sip:179899101234@192.168.0.68;tag=as65768d12
To: sip:85510608707@xxx.xxx.xxx.xx
Call-ID: 6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060
CSeq: 103 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:xxx.xxx.xxx.xx:5060 —>
SIP/2.0 183 Session Progress
Date: Sat, 21 Jan 2012 01:00:03 GMT
Call-ID: 6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060
CSeq: 103 INVITE
Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,PRACK,UPDATE,REFER,SUBSCRIBE,NOTIFY,INFO,REGISTER
Allow-Events: telephone-event
Content-Disposition: session;handling=required
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK621ca390;rport
To: sip:85510608707@xxx.xxx.xxx.xx;tag=36DE86D8-D50
From: “6640757506” sip:179899101234@192.168.0.68;tag=as65768d12
Contact: sip:85510608707@xxx.xxx.xxx.xx:5060
Remote-Party-ID: sip:85510608707@xxx.xxx.xxx.xx:5060;screen=yes;party=calling;privacy=off
Content-Length: 267

v=0
o=CiscoSystemsSIP-GW-UserAgent 3980 6775 IN IP4 119.82.248.17
s=SIP Call
c=IN IP4 119.82.248.17
t=0 0
m=audio 27820 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
c=IN IP4 119.82.248.17
m=video 27822 RTP/AVP 34
c=IN IP4 119.82.248.17
<------------->
— (14 headers 12 lines) —
Found RTP audio format 18
Found audio description format G729 for ID 18
Found RTP video format 34
Capabilities: us - 0x180100 (g729|h263|h263p), peer - audio=0x100 (g729)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80100 (g729|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 119.82.248.17:27820
Peer video RTP is at port 119.82.248.17:27822
– SIP/DATA4ICT-0000002d is making progress passing it to SIP/6640757506-0000002c
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 192.168.1.66:13438 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-f84ea6250b0e840f-1—d8754z-;received=192.168.1.66;rport=13438
From: "6640757506"sip:6640757506@192.168.0.68;tag=e61f5e13
To: "85510608707"sip:85510608707@192.168.0.68;tag=as414f2591
Call-ID: YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:85510608707@192.168.0.68:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 855612938 855612938 IN IP4 192.168.0.68
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.68
t=0 0
m=audio 16876 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:xxx.xxx.xxx.xx:5060 —>
SIP/2.0 200 OK
Date: Sat, 21 Jan 2012 01:00:03 GMT
Call-ID: 6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060
CSeq: 103 INVITE
Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,PRACK,UPDATE,REFER,SUBSCRIBE,NOTIFY,INFO,REGISTER
Allow-Events: telephone-event
Supported: replaces
Content-Disposition: session;handling=required
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK621ca390;rport
To: sip:85510608707@xxx.xxx.xxx.xx;tag=36DE86D8-D50
From: “6640757506” sip:179899101234@192.168.0.68;tag=as65768d12
Contact: sip:85510608707@xxx.xxx.xxx.xx:5060
Remote-Party-ID: sip:85510608707@xxx.xxx.xxx.xx:5060;screen=yes;party=calling;privacy=off
Content-Length: 267

v=0
o=CiscoSystemsSIP-GW-UserAgent 3980 6775 IN IP4 119.82.248.17
s=SIP Call
c=IN IP4 119.82.248.17
t=0 0
m=audio 27820 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
c=IN IP4 119.82.248.17
m=video 27822 RTP/AVP 34
c=IN IP4 119.82.248.17
<------------->
— (15 headers 12 lines) —
list_route: hop: sip:85510608707@xxx.xxx.xxx.xx:5060
set_destination: Parsing sip:85510608707@xxx.xxx.xxx.xx:5060 for address/port to send to
set_destination: set destination to xxx.xxx.xxx.xx:5060
Transmitting (NAT) to xxx.xxx.xxx.xx:5060:
ACK sip:85510608707@xxx.xxx.xxx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK0ab90a6e;rport
Max-Forwards: 70
From: “6640757506” sip:179899101234@192.168.0.68;tag=as65768d12
To: sip:85510608707@xxx.xxx.xxx.xx;tag=36DE86D8-D50
Contact: sip:179899101234@192.168.0.68:5060
Call-ID: 6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.8.1
Content-Length: 0

-- SIP/DATA4ICT-0000002d answered SIP/6640757506-0000002c

Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.1.66:13438 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-f84ea6250b0e840f-1—d8754z-;received=192.168.1.66;rport=13438
From: "6640757506"sip:6640757506@192.168.0.68;tag=e61f5e13
To: "85510608707"sip:85510608707@192.168.0.68;tag=as414f2591
Call-ID: YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
Contact: sip:85510608707@192.168.0.68:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 855612938 855612939 IN IP4 192.168.0.68
s=Asterisk PBX 1.8.8.1
c=IN IP4 192.168.0.68
t=0 0
m=audio 16876 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:192.168.1.66:13438 —>
ACK sip:85510608707@192.168.0.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-857d4943bb7c193b-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:6640757506@192.168.1.66:13438
To: "85510608707"sip:85510608707@192.168.0.68;tag=as414f2591
From: “6640757506"sip:6640757506@192.168.0.68;tag=e61f5e13
Call-ID: YzZjOWE4ODdiY2EwODc1NzIxNzI2NzU3MDlmYTc2OGQ.
CSeq: 2 ACK
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“6640757506”,realm=“asterisk”,nonce=“3e4dc1e1”,uri="sip:85510608707@192.168.0.68”,response=“cfb177ccff4ec1883a82357c34f28f6e”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘2553d87252ac6dfa3ea9e6b8004b006e@192.168.0.68:5060’ Method: BYE
Really destroying SIP dialog ‘NjcyMDFkZWM1OWE1YjIyYzgwZTM5NTM2N2ZlNGMwOWM.’ Method: REGISTER
Really destroying SIP dialog ‘MWRmMzVmZjIzODczNzBiZDc1NDgzYTZmZGQzMDcwNGU.’ Method: BYE
Really destroying SIP dialog ‘3ae5d900629e0b644d984a940050da45@127.0.0.1’ Method: REGISTER

<— SIP read from UDP:192.168.1.66:13438 —>

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

<— SIP read from UDP:192.168.1.66:13438 —>
SUBSCRIBE sip:6640757506@192.168.0.68 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-b65eab4742283670-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:6640757506@192.168.1.66:13438
To: "6640757506"sip:6640757506@192.168.0.68
From: "6640757506"sip:6640757506@192.168.0.68;tag=1277155d
Call-ID: NDQ4NzM4NjYxYzBkZDliMDEyZjFlY2UxYjliZDAwYTI.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1102q stamp 51814
Event: message-summary
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Creating new subscription
Sending to 192.168.1.66:13438 (NAT)
list_route: hop: sip:6640757506@192.168.1.66:13438
Found peer ‘6640757506’ for ‘6640757506’ from 192.168.1.66:13438

<— Transmitting (NAT) to 192.168.1.66:13438 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-b65eab4742283670-1—d8754z-;received=192.168.1.66;rport=13438
From: "6640757506"sip:6640757506@192.168.0.68;tag=1277155d
To: "6640757506"sip:6640757506@192.168.0.68;tag=as76887dc8
Call-ID: NDQ4NzM4NjYxYzBkZDliMDEyZjFlY2UxYjliZDAwYTI.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4d9332e4"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘NDQ4NzM4NjYxYzBkZDliMDEyZjFlY2UxYjliZDAwYTI.’ in 32000 ms (Method: SUBSCRIBE)

<— SIP read from UDP:192.168.1.66:13438 —>
SUBSCRIBE sip:6640757506@192.168.0.68 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-2b57c91c5d3f7a28-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:6640757506@192.168.1.66:13438
To: "6640757506"sip:6640757506@192.168.0.68
From: “6640757506"sip:6640757506@192.168.0.68;tag=1277155d
Call-ID: NDQ4NzM4NjYxYzBkZDliMDEyZjFlY2UxYjliZDAwYTI.
CSeq: 2 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“6640757506”,realm=“asterisk”,nonce=“4d9332e4”,uri="sip:6640757506@192.168.0.68”,response=“cb2b572c5a9cee36f1d9df398b136c1b”,algorithm=MD5
Event: message-summary
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 192.168.1.66:13438 (NAT)
Found peer ‘6640757506’ for ‘6640757506’ from 192.168.1.66:13438

<— Transmitting (NAT) to 192.168.1.66:13438 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 192.168.1.66:13438;branch=z9hG4bK-d8754z-2b57c91c5d3f7a28-1—d8754z-;received=192.168.1.66;rport=13438
From: "6640757506"sip:6640757506@192.168.0.68;tag=1277155d
To: "6640757506"sip:6640757506@192.168.0.68;tag=as76887dc8
Call-ID: NDQ4NzM4NjYxYzBkZDliMDEyZjFlY2UxYjliZDAwYTI.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[Jan 21 08:00:02] NOTICE[14182]: chan_sip.c:24257 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 6640757506
Really destroying SIP dialog ‘NDQ4NzM4NjYxYzBkZDliMDEyZjFlY2UxYjliZDAwYTI.’ Method: SUBSCRIBE

<— SIP read from UDP:xxx.xxx.xxx.xx:5060 —>
OPTIONS sip:179899101234@yyy.yyy.yyy.yy:40414 SIP/2.0
Call-ID: uemSipProxy1327107636133-82159@PROXY.com
CSeq: 1 OPTIONS
From: sip:anonymous@xxx.xxx.xxx.xx:5060;tag=abcdef
To: sip:179899101234@yyy.yyy.yyy.yy:40414
Via: SIP/2.0/UDP xxx.xxx.xxx.xx:5060;branch=1234567890
Max-Forwards: 10
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Looking for 179899101234 in default (domain yyy.yyy.yyy.yy:40414)

<— Transmitting (NAT) to xxx.xxx.xxx.xx:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP xxx.xxx.xxx.xx:5060;branch=1234567890;received=xxx.xxx.xxx.xx;rport=5060
From: sip:anonymous@xxx.xxx.xxx.xx:5060;tag=abcdef
To: sip:179899101234@yyy.yyy.yyy.yy:40414;tag=as65ae3f9b
Call-ID: uemSipProxy1327107636133-82159@PROXY.com
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'uemSipProxy1327107636133-82159@PROXY.com’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:192.168.1.66:13438 —>

<------------->
Really destroying SIP dialog 'uemSipProxy1327107636133-82159@PROXY.com’ Method: OPTIONS

<— SIP read from UDP:192.168.1.66:13438 —>

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

<— SIP read from UDP:xxx.xxx.xxx.xx:5060 —>
BYE sip:179899101234@192.168.0.68:5060 SIP/2.0
Call-ID: 6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060
CSeq: 104 BYE
From: sip:85510608707@xxx.xxx.xxx.xx;tag=36DE86D8-D50
To: “6640757506” sip:179899101234@192.168.0.68;tag=as65768d12
Via: SIP/2.0/UDP xxx.xxx.xxx.xx:5060;branch=z9hG4bK1327107603552
Max-Forwards: 50
Reason: SIP;cause=200;text="Call Snapped"
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to xxx.xxx.xxx.xx:5060 (NAT)
Scheduling destruction of SIP dialog ‘6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060’ in 32000 ms (Method: BYE)

<— Transmitting (NAT) to xxx.xxx.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xx:5060;branch=z9hG4bK1327107603552;received=xxx.xxx.xxx.xx;rport=5060
From: sip:85510608707@xxx.xxx.xxx.xx;tag=36DE86D8-D50
To: “6640757506” sip:179899101234@192.168.0.68;tag=as65768d12
Call-ID: 6b85aaa94ce48ddc53062cf6349f296b@192.168.0.68:5060
CSeq: 104 BYE
Server: Asterisk PBX 1.8.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
– <SIP/6640757506-0000002c> Playing ‘prepaid-enter-dest.gsm’ (language ‘en’)
asteriskCLI> sip set debug off
SIP Debugging Disabled
– <SIP/6640757506-0000002c> Playing ‘prepaid-enter-dest.gsm’ (language ‘en’)
asteriskCLI>

Regards,
Sokphak

6

Definitely being terminated by the Cisco, not by Asterisk. Maybe it ran out of G.729 licences?

7

But if I connect directly by using softphone, it never has this problem. And one more if I use freeswitch, it also never has this problem. So I suppose sth wrong with my asterisk configure or codec negotiation problem. However, the problem also happen when i use alaw&ulaw.

8

Only the Cisco knows why it is dropping the call.

(I’m assuming that you have two way audio, so the problem doesn’t lie with RTP timeouts.)

9

Thanks for prompt respond. But everytime after the call disconnect when I tried to call again it said “Everyone is busy/congested at this time (1:0/1/0)”. After I reload in CLI mode, it available to call again.

Regards,
Sokphak

10

Everyone… is a very non-specific error. You need to get the detailed traces, including any sip set debug ones, to see why it is actually being produced. sip show peers may be instructive.

I would also stress that you need to look at the BYE from Cisco side.

At the moment, I can only guess that a keepalive (qualify), or register, has timed out.

11

One detail that I missed is that it is the caller that is dropping the call, not the callee. The SIP trace doesn’t show the outgoing call being cleared, which might explain the subsequent busy. If so, that could be a problem with the AGI handling, but there is no tracing of that.

You still need to work out why the caller is dropping the call.

12

So, could you please advice what is the command or method to identified the exact problem?

13

I’m not familiar with the logging capabilities of the Cisco gateway.

You should run sip set debug on, verbose 3, and probably sip set debug 5 for chan_sip, for the two calls. You should also use “sip show peers” between the calls.

14

I still cannot find any solution for this problem. It keep disconnect call every 90s to 110s. I try many option from this forum. Could you please help more?

Regards,
Sokphak