SipXtapi Dropped Call Issue (Urgent) Help Needed...

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Ramkumar Sridharan

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Oct 7, 2014, 11:49:12 PM10/7/14

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We are facing voice quality issues and also dropped call issues with our SipXtapi implementation.... Close to 10% of the calls are dropped...

I have provided the log information from SipX below for your reference and also a wireshark pcap file for the call flow..

This is for a inbound scenario, where the dialer sends a 100 and later sends a cancel after 12 seconds of wait time...

We have been trying to identify the root cause but to no available...

SipXtapi v3.2.0

Please help..

See below some of the log entries...

"2014-10-06T17:58:18.222000":16137:INCOMING_PARSED:DEBUG:qorcwlchft21:SipClient-33:00000CC0:sipXtapi:"INVITE sip:70034...@10.4.152.59:5060;LINEID=14313c07926a SIP/2.0\r\nMax-Forwards: 70\r\nFrom: <sip:341...@10.3.36.49>;tag=as25dc6134\r\nTo: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>\r\nContact: <sip:341...@10.3.36.49:5060>\r\nCall-Id: 65bfff4576bf6e9b...@10.3.36.49:5060\r\nCseq: 102 INVITE\r\nUser-Agent: Asterisk PBX 11.7.0\r\nDate: Mon, 06 Oct 2014 17:58:10 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\r\nSupported: replaces, timer\r\nContent-Type: application/sdp\r\nContent-Length: 229\r\nVia: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport=5060\r\n\r\nv=0\r\no=root 971436356 971436356 IN IP4 10.3.36.49\r\ns=Asterisk PBX 11.7.0\r\nc=IN IP4 10.3.36.49\r\nt=0 0\r\nm=audio 13534 RTP/AVP 0 101\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n++++++++++++++++++++END++++++++++++++++++++\n"

"2014-10-06T17:58:18.223000":16138:CP:WARNING:qorcwlchft21:CallManager-44:000011B8:sipXtapi:"willHandleMessage: Ignoring SIP request for dropping call: 65bfff4576bf6e9b...@10.3.36.49:5060"

Thanks.

drop call flow.png

Daniel Petrie

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Oct 8, 2014, 1:42:56 AM10/8/14

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Hi Ramkumar:

Just to confirm, you are using the sipX from the svn repository:

http://scm.sipfoundry.org/rep/sipX/main/

Can you send a clean, complete log file with just one call, the failed one, with logging at DEBUG level.

What is the server and end point on the other side?

My first guess is that there is something wrong with the transaction data in the 100 response such that it does not match the INVITE, but I would need to look at the log and details.

Cheers,

Dan

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Ramkumar Sridharan

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Oct 8, 2014, 9:12:53 AM10/8/14

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Hi Dan,

We are using the sipx 3.2.0 version from sipfoundry...

I have attached the log for 1 dropped call..

SipXtapi registers with Asterisk Server on the other side.

Please let me know your thoughts...

Thanks a lot.

On Wednesday, 8 October 2014 11:12:56 UTC+5:30, Dan Petrie wrote:

Hi Ramkumar:
Just to confirm, you are using the sipX from the svn repository:
http://scm.sipfoundry.org/rep/sipX/main/

Can you send a clean, complete log file with just one call, the failed one, with logging at DEBUG level.

What is the server and end point on the other side?

My first guess is that there is something wrong with the transaction data in the 100 response such that it does not match the INVITE, but I would need to look at the log and details.

Cheers,
Dan

On Tuesday, October 7, 2014 11:49 PM, Ramkumar Sridharan <ramkumar....@gmail.com> wrote:

We are facing voice quality issues and also dropped call issues with our SipXtapi implementation.... Close to 10% of the calls are dropped...

I have provided the log information from SipX below for your reference and also a wireshark pcap file for the call flow..

This is for a inbound scenario, where the dialer sends a 100 and later sends a cancel after 12 seconds of wait time...

We have been trying to identify the root cause but to no available...

SipXtapi v3.2.0

Please help..

See below some of the log entries...

"2014-10-06T17:58:18.222000":16137:INCOMING_PARSED:DEBUG:qorcwlchft21:SipClient-33:00000CC0:sipXtapi:"INVITE sip:70034...@10.4.152.59:5060;LINEID=14313c07926a SIP/2.0\r\nMax-Forwards: 70\r\nFrom: <sip:3...@10.3.36.49>;tag=as25dc6134\r\nTo: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>\r\nContact: <sip:341...@10.3.36.49:5060>\r\nCall-Id: 65bfff4576bf6e9b53f067a07265b7f...@10.3.36.49:5060\r\nCseq: 102 INVITE\r\nUser-Agent: Asterisk PBX 11.7.0\r\nDate: Mon, 06 Oct 2014 17:58:10 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\r\nSupported: replaces, timer\r\nContent-Type: application/sdp\r\nContent-Length: 229\r\nVia: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport=5060\r\n\r\nv=0\r\no=root 971436356 971436356 IN IP4 10.3.36.49\r\ns=Asterisk PBX 11.7.0\r\nc=IN IP4 10.3.36.49\r\nt=0 0\r\nm=audio 13534 RTP/AVP 0 101\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n++++++++++++++++++++END++++++++++++++++++++\n"

"2014-10-06T17:58:18.223000":16138:CP:WARNING:qorcwlchft21:CallManager-44:000011B8:sipXtapi:"willHandleMessage: Ignoring SIP request for dropping call: 65bfff4576bf6e9b53f067a07265b7f...@10.3.36.49:5060"

Thanks.

sipxtapi one failure call log.txt

Daniel Petrie

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Oct 8, 2014, 11:08:17 AM10/8/14

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Hi Ramkumar:

I have had a look at the messages in the log and I cannot see anything wrong with the 100 Trying response. The transaction data in the 100 Trying appears to be identical to the INVITE. So I can see no reason that the Asterisk box should not match the 100 response to the invite.

I would have expected that you would have sent 180 response from sipX sooner though. In order for sipX to emit a 180, you need to call the sipxCallAccept method.

I am not an Asterisk expert, but I would guess that the Asterisk box is setup to cancel if it does not see a 180 or 200 within some time out period. You can either try to figure out how to configure the Asterisk to have a longer time out or you need to call sipxCallAccept sooner.

Cheers,

Dan

FYI, here is the message flow that I saw:

2014-10-06T17:58:18.222000

incoming

INVITE sip:70034...@10.4.152.59:5060;LINEID=14313c07926a SIP/2.0

Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport

Max-Forwards: 70

From: <sip:341...@10.3.36.49>;tag=as25dc6134

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>

Contact: <sip:341...@10.3.36.49:5060>

Call-ID: 65bfff4576bf6e9b...@10.3.36.49:5060

CSeq: 102 INVITE

User-Agent: Asterisk PBX 11.7.0

Date: Mon, 06 Oct 2014 17:58:10 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 229

v=0

o=root 971436356 971436356 IN IP4 10.3.36.49

s=Asterisk PBX 11.7.0

c=IN IP4 10.3.36.49

t=0 0

m=audio 13534 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

2014-10-06T17:58:18.221000

SIP/2.0 100 Trying

From: <sip:341...@10.3.36.49>;tag=as25dc6134

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>

Call-Id: 65bfff4576bf6e9b...@10.3.36.49:5060

Cseq: 102 INVITE

Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport=5060

Contact: <sip:10.4.152.59:5060>

Content-Length: 0

2014-10-06T17:58:25.643000

outgoing:

2014-10-06T17:58:25.666000

incoming:

2014-10-06T17:58:25.685000

outgoing:

2014-10-06T17:58:25.690000

incoming:

2014-10-06T17:58:25.710000

incoming:

OPTIONS

2014-10-06T17:58:25.719000

Outgoing:

SIP/2.0 200 OK

From: "asterisk" <sip:aste...@10.3.36.49>;tag=as7753c57a

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>

Call-Id: 5f3710046089ff01...@10.3.36.49:5060

Cseq: 102 OPTIONS

Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK0dcffc7f;rport=5060

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY

Supported: replaces

Contact: <sip:10.4.152.59:5060>

Content-Length: 0

2014-10-06T17:58:25.721000

Incoming:

2014-10-06T17:58:29.904000

Incoming:

CANCEL sip:70034...@10.4.152.59:5060;LINEID=14313c07926a SIP/2.0

Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport

Max-Forwards: 70

From: <sip:341...@10.3.36.49>;tag=as25dc6134

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>

Call-ID: 65bfff4576bf6e9b...@10.3.36.49:5060

CSeq: 102 CANCEL

User-Agent: Asterisk PBX 11.7.0

Content-Length: 0

2014-10-06T17:58:45.745000

Outgoing:

2014-10-06T17:58:45.767000

Incoming:

2014-10-06T17:58:45.811000

Incoming:

OPTIONS

2014-10-06T17:58:45.824000

Incoming:

On Wednesday, October 8, 2014 9:12 AM, Ramkumar Sridharan <ramkumar....@gmail.com> wrote:

Hi Dan,

We are using the sipx 3.2.0 version from sipfoundry...

I have attached the log for 1 dropped call..

SipXtapi registers with Asterisk Server on the other side.

Please let me know your thoughts...

Thanks a lot.

On Wednesday, 8 October 2014 11:12:56 UTC+5:30, Dan Petrie wrote:

Hi Ramkumar:
Just to confirm, you are using the sipX from the svn repository:
http://scm.sipfoundry.org/rep/ sipX/main/

Can you send a clean, complete log file with just one call, the failed one, with logging at DEBUG level.

What is the server and end point on the other side?

My first guess is that there is something wrong with the transaction data in the 100 response such that it does not match the INVITE, but I would need to look at the log and details.

Cheers,
Dan

On Tuesday, October 7, 2014 11:49 PM, Ramkumar Sridharan <ramkumar....@gmail.com> wrote:

We are facing voice quality issues and also dropped call issues with our SipXtapi implementation.... Close to 10% of the calls are dropped...

I have provided the log information from SipX below for your reference and also a wireshark pcap file for the call flow..

This is for a inbound scenario, where the dialer sends a 100 and later sends a cancel after 12 seconds of wait time...

We have been trying to identify the root cause but to no available...

SipXtapi v3.2.0

Please help..

See below some of the log entries...

"2014-10-06T17:58:18.222000": 16137:INCOMING_PARSED:DEBUG: qorcwlchft21:SipClient-33: 00000CC0:sipXtapi:"INVITE sip:70034...@10.4.152.59: 5060;LINEID=14313c07926a SIP/2.0\r\nMax-Forwards: 70\r\nFrom: <sip:3...@10.3.36.49>;tag= as25dc6134\r\nTo: <sip:70034...@10.4.152.59: 5060;LINEID=14313c07926a>\r\ nContact: <sip:341...@10.3.36.49:5060>\ r\nCall-Id: 65bfff4576bf6e9b53f067a07265b7 f...@10.3.36.49:5060\r\nCseq: 102 INVITE\r\nUser-Agent: Asterisk PBX 11.7.0\r\nDate: Mon, 06 Oct 2014 17:58:10 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\r\nSupported: replaces, timer\r\nContent-Type: application/sdp\r\nContent- Length: 229\r\nVia: SIP/2.0/UDP 10.3.36.49:5060;branch= z9hG4bK154c227a;rport=5060\r\ n\r\nv=0\r\no=root 971436356 971436356 IN IP4 10.3.36.49\r\ns=Asterisk PBX 11.7.0\r\nc=IN IP4 10.3.36.49\r\nt=0 0\r\nm=audio 13534 RTP/AVP 0 101\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na= fmtp:101 0-16\r\na=ptime:20\r\na= sendrecv\r\n++++++++++++++++++ ++END++++++++++++++++++++\n"

"2014-10-06T17:58:18.223000": 16138:CP:WARNING:qorcwlchft21: CallManager-44:000011B8: sipXtapi:"willHandleMessage: Ignoring SIP request for dropping call: 65bfff4576bf6e9b53f067a07265b7 f...@10.3.36.49:5060"

Thanks.

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Ramkumar Sridharan

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Oct 9, 2014, 3:15:44 AM10/9/14

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Hi Dan,

Thanks for your response again...

We are calling the sipxCallAccept method and also understand the asterisk time out is quit esufficient.

For some reason the 180 is not sent by the application ... it was not able to get to this and it was waiting for some reason (for the 10% of the scenarios) ... Which puzzles us as to what possible scenarios could ccause this...

To put better context, I am attaching a zip file (log file) which contain previous call and the dropped call.. It appears the dropped call resumed after the old call got cleaned up...

I will put this here so you can indedpently look at it and advise us...

Please let me know.

Thanks a lot.

On Wednesday, 8 October 2014 20:38:17 UTC+5:30, Dan Petrie wrote:

Hi Ramkumar:

I have had a look at the messages in the log and I cannot see anything wrong with the 100 Trying response. The transaction data in the 100 Trying appears to be identical to the INVITE. So I can see no reason that the Asterisk box should not match the 100 response to the invite.

I would have expected that you would have sent 180 response from sipX sooner though. In order for sipX to emit a 180, you need to call the sipxCallAccept method.

I am not an Asterisk expert, but I would guess that the Asterisk box is setup to cancel if it does not see a 180 or 200 within some time out period. You can either try to figure out how to configure the Asterisk to have a longer time out or you need to call sipxCallAccept sooner.

Cheers,
Dan

FYI, here is the message flow that I saw:

2014-10-06T17:58:18.222000
incoming
INVITE sip:70034...@10.4.152.59:5060;LINEID=14313c07926a SIP/2.0
Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport
Max-Forwards: 70

From: <sip:3...@10.3.36.49>;tag=as25dc6134

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>
Contact: <sip:341...@10.3.36.49:5060>

Call-ID: 65bfff4576bf6e9b53f067a07265b7f...@10.3.36.49:5060

CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Mon, 06 Oct 2014 17:58:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 971436356 971436356 IN IP4 10.3.36.49
s=Asterisk PBX 11.7.0
c=IN IP4 10.3.36.49
t=0 0
m=audio 13534 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2014-10-06T17:58:18.221000
SIP/2.0 100 Trying

From: <sip:3...@10.3.36.49>;tag=as25dc6134

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>

Call-Id: 65bfff4576bf6e9b53f067a07265b7f...@10.3.36.49:5060

Cseq: 102 INVITE
Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport=5060
Contact: <sip:10.4.152.59:5060>
Content-Length: 0

2014-10-06T17:58:25.643000
outgoing:
REGISTER

2014-10-06T17:58:25.666000
incoming:
REGISTER 401 Unauth

2014-10-06T17:58:25.685000
outgoing:
REGISTER

2014-10-06T17:58:25.690000
incoming:
REGISTER 401 unauth

2014-10-06T17:58:25.710000
incoming:
OPTIONS

2014-10-06T17:58:25.719000
Outgoing:
SIP/2.0 200 OK

From: "asterisk" <sip:as...@10.3.36.49>;tag=as7753c57a

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>

Call-Id: 5f3710046089ff010f8d687b20a82f6...@10.3.36.49:5060

Cseq: 102 OPTIONS
Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK0dcffc7f;rport=5060
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY
Supported: replaces
Contact: <sip:10.4.152.59:5060>
Content-Length: 0

2014-10-06T17:58:25.721000
Incoming:
REGISTER 200 Ok

2014-10-06T17:58:29.904000
Incoming:
CANCEL sip:70034...@10.4.152.59:5060;LINEID=14313c07926a SIP/2.0
Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport
Max-Forwards: 70

From: <sip:3...@10.3.36.49>;tag=as25dc6134

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>

Call-ID: 65bfff4576bf6e9b53f067a07265b7f...@10.3.36.49:5060

"2014-10-06T17:58:18.222000": 16137:INCOMING_PARSED:DEBUG: qorcwlchft21:SipClient-33: 00000CC0:sipXtapi:"INVITE sip:70034...@10.4.152.59: 5060;LINEID=14313c07926a SIP/2.0\r\nMax-Forwards: 70\r\nFrom: <sip:3...@10.3.36.49>;tag= as25dc6134\r\nTo: <sip:700...@10.4.152.59: 5060;LINEID=14313c07926a>\r\ nContact: <sip:341...@10.3.36.49:5060>\ r\nCall-Id: 65bfff4576bf6e9b53f067a07265b7 f...@10.3.36.49:5060\r\nCseq: 102 INVITE\r\nUser-Agent: Asterisk PBX 11.7.0\r\nDate: Mon, 06 Oct 2014 17:58:10 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\r\nSupported: replaces, timer\r\nContent-Type: application/sdp\r\nContent- Length: 229\r\nVia: SIP/2.0/UDP 10.3.36.49:5060;branch= z9hG4bK154c227a;rport=5060\r\ n\r\nv=0\r\no=root 971436356 971436356 IN IP4 10.3.36.49\r\ns=Asterisk PBX 11.7.0\r\nc=IN IP4 10.3.36.49\r\nt=0 0\r\nm=audio 13534 RTP/AVP 0 101\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na= fmtp:101 0-16\r\na=ptime:20\r\na= sendrecv\r\n++++++++++++++++++ ++END++++++++++++++++++++\n"

"2014-10-06T17:58:18.223000": 16138:CP:WARNING:qorcwlchft21: CallManager-44:000011B8: sipXtapi:"willHandleMessage: Ignoring SIP request for dropping call: 65bfff4576bf6e9b53f067a07265b7 f...@10.3.36.49:5060"

Thanks.

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Problem_Call_Context.zip

Ramkumar Sridharan

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Oct 9, 2014, 1:19:33 PM10/9/14

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Hi Dan,

The issue is very critical with our client and need help urgently.. Is there a way we can reach you and/or get more direct support on this...

Please let me know...

Thanks..

On Wednesday, 8 October 2014 20:38:17 UTC+5:30, Dan Petrie wrote:

Hi Ramkumar:

I have had a look at the messages in the log and I cannot see anything wrong with the 100 Trying response. The transaction data in the 100 Trying appears to be identical to the INVITE. So I can see no reason that the Asterisk box should not match the 100 response to the invite.

I would have expected that you would have sent 180 response from sipX sooner though. In order for sipX to emit a 180, you need to call the sipxCallAccept method.

I am not an Asterisk expert, but I would guess that the Asterisk box is setup to cancel if it does not see a 180 or 200 within some time out period. You can either try to figure out how to configure the Asterisk to have a longer time out or you need to call sipxCallAccept sooner.

Cheers,
Dan

FYI, here is the message flow that I saw:

2014-10-06T17:58:18.222000
incoming
INVITE sip:70034...@10.4.152.59:5060;LINEID=14313c07926a SIP/2.0
Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport
Max-Forwards: 70

From: <sip:3...@10.3.36.49>;tag=as25dc6134

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>
Contact: <sip:341...@10.3.36.49:5060>

Call-ID: 65bfff4576bf6e9b53f067a07265b7f...@10.3.36.49:5060

CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Mon, 06 Oct 2014 17:58:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 971436356 971436356 IN IP4 10.3.36.49
s=Asterisk PBX 11.7.0
c=IN IP4 10.3.36.49
t=0 0
m=audio 13534 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2014-10-06T17:58:18.221000
SIP/2.0 100 Trying

From: <sip:3...@10.3.36.49>;tag=as25dc6134

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>

Call-Id: 65bfff4576bf6e9b53f067a07265b7f...@10.3.36.49:5060

Cseq: 102 INVITE
Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport=5060
Contact: <sip:10.4.152.59:5060>
Content-Length: 0

2014-10-06T17:58:25.643000
outgoing:
REGISTER

2014-10-06T17:58:25.666000
incoming:
REGISTER 401 Unauth

2014-10-06T17:58:25.685000
outgoing:
REGISTER

2014-10-06T17:58:25.690000
incoming:
REGISTER 401 unauth

2014-10-06T17:58:25.710000
incoming:
OPTIONS

2014-10-06T17:58:25.719000
Outgoing:
SIP/2.0 200 OK

From: "asterisk" <sip:as...@10.3.36.49>;tag=as7753c57a

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>

Call-Id: 5f3710046089ff010f8d687b20a82f6...@10.3.36.49:5060

Cseq: 102 OPTIONS
Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK0dcffc7f;rport=5060
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY
Supported: replaces
Contact: <sip:10.4.152.59:5060>
Content-Length: 0

2014-10-06T17:58:25.721000
Incoming:
REGISTER 200 Ok

2014-10-06T17:58:29.904000
Incoming:
CANCEL sip:70034...@10.4.152.59:5060;LINEID=14313c07926a SIP/2.0
Via: SIP/2.0/UDP 10.3.36.49:5060;branch=z9hG4bK154c227a;rport
Max-Forwards: 70

From: <sip:3...@10.3.36.49>;tag=as25dc6134

To: <sip:70034...@10.4.152.59:5060;LINEID=14313c07926a>

Call-ID: 65bfff4576bf6e9b53f067a07265b7f...@10.3.36.49:5060

"2014-10-06T17:58:18.222000": 16137:INCOMING_PARSED:DEBUG: qorcwlchft21:SipClient-33: 00000CC0:sipXtapi:"INVITE sip:70034...@10.4.152.59: 5060;LINEID=14313c07926a SIP/2.0\r\nMax-Forwards: 70\r\nFrom: <sip:3...@10.3.36.49>;tag= as25dc6134\r\nTo: <sip:700...@10.4.152.59: 5060;LINEID=14313c07926a>\r\ nContact: <sip:341...@10.3.36.49:5060>\ r\nCall-Id: 65bfff4576bf6e9b53f067a07265b7 f...@10.3.36.49:5060\r\nCseq: 102 INVITE\r\nUser-Agent: Asterisk PBX 11.7.0\r\nDate: Mon, 06 Oct 2014 17:58:10 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\r\nSupported: replaces, timer\r\nContent-Type: application/sdp\r\nContent- Length: 229\r\nVia: SIP/2.0/UDP 10.3.36.49:5060;branch= z9hG4bK154c227a;rport=5060\r\ n\r\nv=0\r\no=root 971436356 971436356 IN IP4 10.3.36.49\r\ns=Asterisk PBX 11.7.0\r\nc=IN IP4 10.3.36.49\r\nt=0 0\r\nm=audio 13534 RTP/AVP 0 101\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na= fmtp:101 0-16\r\na=ptime:20\r\na= sendrecv\r\n++++++++++++++++++ ++END++++++++++++++++++++\n"

"2014-10-06T17:58:18.223000": 16138:CP:WARNING:qorcwlchft21: CallManager-44:000011B8: sipXtapi:"willHandleMessage: Ignoring SIP request for dropping call: 65bfff4576bf6e9b53f067a07265b7 f...@10.3.36.49:5060"

Thanks.

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