Retransmission timeout reached

rg25 · August 10, 2014, 7:52pm

I am getting this error when trying to call from user5 to user6. Though the sip handshake succeeds (ringing on both sides, phone is being picked up), the media transmission never takes place. The server issues 489 bad event after receiving UA picks up the call.

My topology is …

user5 is registered via android phone zoiper application.
user6 is registered via blink softphone behind nat firewall. All outgoing ports are open.
asterisk server is sitting on a remote wan machine behind nat firewall. All incoming and outgoing udp ports are open.

Asterisk 11.9

sip.conf

[user5]
type=friend
defaultuser=user5
allow=h263p
secret=1234
host=dynamic
canreinvite=no
context=default
nat=force_rport
insecure=port,invite
qualify=yes

[user6]
type=friend
defaultuser=user6
allow=h263p
secret=1234
host=dynamic
canreinvite=no
context=default
nat=force_rport
insecure=port,invite
qualify=yes

extensions.conf

exten => _user.,1,Dial(SIP/${EXTEN})

Thanks in advance

ambiorixg12 · August 11, 2014, 4:02am

SIP trace is needed

david55 · August 11, 2014, 10:07am

Why do you want anyone who knows the names user4 and user5 to be able to make unauthenticated calls?

I suspect there is a NAT involvement here, so, as well as the SIP traces, you need to provide network topology information.

rg25 · August 11, 2014, 1:45pm

III.III.II - itsp wan subnet
AA.AAA.AAA - asterisk wan subnet
SS.SS.SSS - user6 user agent wan subnet
CCC.CCC.CCC - user5 user agent wan subnet (android using att wireless network)
192.168.1.126 - asterisk lan ip
192.168.7.132 - user6 user agent lan ip

user6 ua is sitting behind nat firewall and registered to asterisk server sitting behind nat firewall. user5 ua is sitting on att wireless data network registered to the same asterisk server. This asterisk server is registered to sip trunk provider.

rg25 · August 11, 2014, 1:46pm

[code]<— SIP read from UDP:III.III.II.144:5060 —>
OPTIONS sip:AA.AAA.AAA.201:5060 SIP/2.0
Max-Forwards: 10
Record-Route: sip:III.III.II.144;lr
Via: SIP/2.0/UDP III.III.II.144;branch=z9hG4bK1cb7.7916b5b50c9263440ae1a1517dfd2995.0
Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
Route: sip:III.III.II.144;lr;received='sip:AA.AAA.AAA.201:5060’
From: sip:ping@invalid;tag=d8a0f408
To: sip:AA.AAA.AAA.201:5060
Call-ID: 0c26e475-3fc32907-b869138@70.167.153.136
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to III.III.II.144:5060 (NAT)
Looking for s in default (domain AA.AAA.AAA.201)

<— Transmitting (NAT) to III.III.II.144:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP III.III.II.144;branch=z9hG4bK1cb7.7916b5b50c9263440ae1a1517dfd2995.0;received=III.III.II.144;rport=5060
Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
Record-Route: sip:III.III.II.144;lr
From: sip:ping@invalid;tag=d8a0f408
To: sip:AA.AAA.AAA.201:5060;tag=as040ab73e
Call-ID: 0c26e475-3fc32907-b869138@70.167.153.136
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:192.168.1.126:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0c26e475-3fc32907-b869138@70.167.153.136’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘fb224c2-7e00bbd6-29e6ac6@III.III.II.131’ Method: OPTIONS
Really destroying SIP dialog ‘0c26e475-3fc32907-b869138@70.167.153.136’ Method: OPTIONS

<— SIP read from UDP:III.III.II.144:5060 —>
OPTIONS sip:AA.AAA.AAA.201:5060 SIP/2.0
Max-Forwards: 10
Record-Route: sip:III.III.II.144;lr
Via: SIP/2.0/UDP III.III.II.144;branch=z9hG4bK86f1.193a96d750895560a868a2ef22859d13.0
Via: SIP/2.0/UDP III.III.II.131:5060;branch=0
Route: sip:III.III.II.144;lr;received='sip:AA.AAA.AAA.201:5060’
From: sip:ping@invalid;tag=11c09b25
To: sip:AA.AAA.AAA.201:5060
Call-ID: fb224c2-71f1bbd6-9ce6ac6@III.III.II.131
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to III.III.II.144:5060 (NAT)
Looking for s in default (domain AA.AAA.AAA.201)

<— Transmitting (NAT) to III.III.II.144:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP III.III.II.144;branch=z9hG4bK86f1.193a96d750895560a868a2ef22859d13.0;received=III.III.II.144;rport=5060
Via: SIP/2.0/UDP III.III.II.131:5060;branch=0
Record-Route: sip:III.III.II.144;lr
From: sip:ping@invalid;tag=11c09b25
To: sip:AA.AAA.AAA.201:5060;tag=as0a1b890c
Call-ID: fb224c2-71f1bbd6-9ce6ac6@III.III.II.131
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:192.168.1.126:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘fb224c2-71f1bbd6-9ce6ac6@III.III.II.131’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:CCC.CCC.CCC.59:33869 —>
REGISTER sip:AA.AAA.AAA.201;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.122.63.126:65391;branch=z9hG4bK-d8754z-cb548b7c83d875e9-1—d8754z-
Max-Forwards: 70
Contact: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP
To: sip:user5@AA.AAA.AAA.201;transport=UDP
From: sip:user5@AA.AAA.AAA.201;transport=UDP;tag=8742c84c
Call-ID: ZjlmYWQ4ZjE0YmRlZWE4MTBhNDg1MmE2ZjhkMjdlMWY.
CSeq: 1 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r26025
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to CCC.CCC.CCC.59:33869 (NAT)
Sending to CCC.CCC.CCC.59:33869 (NAT)

<— Transmitting (NAT) to CCC.CCC.CCC.59:33869 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.122.63.126:65391;branch=z9hG4bK-d8754z-cb548b7c83d875e9-1—d8754z-;received=CCC.CCC.CCC.59;rport=33869
From: sip:user5@AA.AAA.AAA.201;transport=UDP;tag=8742c84c
To: sip:user5@AA.AAA.AAA.201;transport=UDP;tag=as74d5bc09
Call-ID: ZjlmYWQ4ZjE0YmRlZWE4MTBhNDg1MmE2ZjhkMjdlMWY.
CSeq: 1 REGISTER
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="13cfb76b"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZjlmYWQ4ZjE0YmRlZWE4MTBhNDg1MmE2ZjhkMjdlMWY.’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:CCC.CCC.CCC.59:33869 —>
REGISTER sip:AA.AAA.AAA.201;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.122.63.126:65391;branch=z9hG4bK-d8754z-94d4ba14ab700c90-1—d8754z-
Max-Forwards: 70
Contact: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP
To: sip:user5@AA.AAA.AAA.201;transport=UDP
From: sip:user5@AA.AAA.AAA.201;transport=UDP;tag=8742c84c
Call-ID: ZjlmYWQ4ZjE0YmRlZWE4MTBhNDg1MmE2ZjhkMjdlMWY.
CSeq: 2 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r26025
Authorization: Digest username=“user5”,realm=“asterisk”,nonce=“13cfb76b”,uri=“sip:AA.AAA.AAA.201;transport=UDP”,response=“ce3f7c9417041f2f6d53128407435b0c”,algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to CCC.CCC.CCC.59:33869 (NAT)
Reliably Transmitting (NAT) to CCC.CCC.CCC.59:33869:
OPTIONS sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK6ee33cb3;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.126;tag=as768aa870
To: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP
Contact: sip:asterisk@192.168.1.126:5060
Call-ID: 1b20b361365731d25f4510141e00000f@192.168.1.126:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.9.0
Date: Mon, 11 Aug 2014 12:41:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— Transmitting (NAT) to CCC.CCC.CCC.59:33869 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.122.63.126:65391;branch=z9hG4bK-d8754z-94d4ba14ab700c90-1—d8754z-;received=CCC.CCC.CCC.59;rport=33869
From: sip:user5@AA.AAA.AAA.201;transport=UDP;tag=8742c84c
To: sip:user5@AA.AAA.AAA.201;transport=UDP;tag=as74d5bc09
Call-ID: ZjlmYWQ4ZjE0YmRlZWE4MTBhNDg1MmE2ZjhkMjdlMWY.
CSeq: 2 REGISTER
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP;expires=60
Date: Mon, 11 Aug 2014 12:41:30 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZjlmYWQ4ZjE0YmRlZWE4MTBhNDg1MmE2ZjhkMjdlMWY.’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:CCC.CCC.CCC.59:33869 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK6ee33cb3;rport=5060;received=AA.AAA.AAA.201
Contact: sip:10.122.63.126:65391
To: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP;tag=756fdb0c
From: "asterisk"sip:asterisk@192.168.1.126;tag=as768aa870
Call-ID: 1b20b361365731d25f4510141e00000f@192.168.1.126:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r26025
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
[Aug 11 08:41:31] NOTICE[40118]: chan_sip.c:23645 handle_response_peerpoke: Peer ‘user5’ is now Reachable. (134ms / 2000ms)
Really destroying SIP dialog ‘1b20b361365731d25f4510141e00000f@192.168.1.126:5060’ Method: OPTIONS

<— SIP read from UDP:III.III.II.144:5060 —>
OPTIONS sip:AA.AAA.AAA.201:5060 SIP/2.0
Max-Forwards: 10
Record-Route: sip:III.III.II.144;lr
Via: SIP/2.0/UDP III.III.II.144;branch=z9hG4bKca56.46217e8f500e8bf06066c98557c1014b.0
Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
Route: sip:III.III.II.144;lr;received='sip:AA.AAA.AAA.201:5060’
From: sip:ping@invalid;tag=cc82f408
To: sip:AA.AAA.AAA.201:5060
Call-ID: 0c26e475-23b52907-2c69138@70.167.153.136
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to III.III.II.144:5060 (NAT)
Looking for s in default (domain AA.AAA.AAA.201)

<— Transmitting (NAT) to III.III.II.144:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP III.III.II.144;branch=z9hG4bKca56.46217e8f500e8bf06066c98557c1014b.0;received=III.III.II.144;rport=5060
Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
Record-Route: sip:III.III.II.144;lr
From: sip:ping@invalid;tag=cc82f408
To: sip:AA.AAA.AAA.201:5060;tag=as550a1977
Call-ID: 0c26e475-23b52907-2c69138@70.167.153.136
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:192.168.1.126:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0c26e475-23b52907-2c69138@70.167.153.136’ in 32000 ms (Method: OPTIONS)
[Aug 11 08:41:44] NOTICE[40118]: chan_sip.c:15106 sip_reregister: – Re-registration for 33330580@sip.itsp.com
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to III.III.II.144:5060:
REGISTER sip:sip.itsp.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK40b73aae;rport
Max-Forwards: 70
From: sip:33330580@sip.itsp.com;tag=as372c9fe5
To: sip:33330580@sip.itsp.com
Call-ID: 5cfab0f96e6357612742df07339ba6c6@67.63.55.3
CSeq: 798 REGISTER
User-Agent: Asterisk PBX 11.9.0
Authorization: Digest username=“33330580”, realm=“sip.itsp.com”, algorithm=MD5, uri=“sip:sip.itsp.com”, nonce=“U+i5+FPouMzUixIgzO4srucoBE4iWWN2”, response=“ab3b4b46aa811f3554e52a8266dd2bd1”, qop=auth, cnonce=“15c7b144”, nc=00000004
Expires: 120
Contact: sip:s@192.168.1.126:5060
Content-Length: 0

<— SIP read from UDP:III.III.II.144:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.126:5060;received=AA.AAA.AAA.201;branch=z9hG4bK40b73aae;rport=5060
From: sip:33330580@sip.itsp.com;tag=as372c9fe5
To: sip:33330580@sip.itsp.com;tag=aa681f9fdf30149b00040f579a1d99c4.601d
Call-ID: 5cfab0f96e6357612742df07339ba6c6@67.63.55.3
CSeq: 798 REGISTER
WWW-Authenticate: Digest realm=“sip.itsp.com”, nonce=“U+i7M1PougeCgRely8N7tFP+gw2uFtcX”, qop="auth"
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Responding to challenge, registration to domain/host name sip.itsp.com
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to III.III.II.144:5060:
REGISTER sip:sip.itsp.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK28deedd9;rport
Max-Forwards: 70
From: sip:33330580@sip.itsp.com;tag=as372c9fe5
To: sip:33330580@sip.itsp.com
Call-ID: 5cfab0f96e6357612742df07339ba6c6@67.63.55.3
CSeq: 799 REGISTER
User-Agent: Asterisk PBX 11.9.0
Authorization: Digest username=“33330580”, realm=“sip.itsp.com”, algorithm=MD5, uri=“sip:sip.itsp.com”, nonce=“U+i7M1PougeCgRely8N7tFP+gw2uFtcX”, response=“ddfc353e28f43959f2862ead5440b4a0”, qop=auth, cnonce=“601ca25a”, nc=00000001
Expires: 120
Contact: sip:s@192.168.1.126:5060
Content-Length: 0

<— SIP read from UDP:III.III.II.144:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.126:5060;received=AA.AAA.AAA.201;branch=z9hG4bK28deedd9;rport=5060
From: sip:33330580@sip.itsp.com;tag=as372c9fe5
To: sip:33330580@sip.itsp.com;tag=aa681f9fdf30149b00040f579a1d99c4.181c
Call-ID: 5cfab0f96e6357612742df07339ba6c6@67.63.55.3
CSeq: 799 REGISTER
Contact: sip:s@192.168.1.126:5060;q=1;expires=120;received=“sip:AA.AAA.AAA.201:5060”, sip:33330580@SS.SS.SSS.253:5080;transport=udp;q=1;expires=371;received="sip:SS.SS.SSS.253:5080"
Content-Length: 0

<------------->
— (8 headers 0 lines) —
[Aug 11 08:41:44] NOTICE[40118]: chan_sip.c:23595 handle_response_register: Outbound Registration: Expiry for sip.itsp.com is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog ‘5cfab0f96e6357612742df07339ba6c6@67.63.55.3’ Method: REGISTER

<— SIP read from UDP:SS.SS.SSS.254:53938 —>
REGISTER sip:AA.AAA.AAA.201 SIP/2.0
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;rport;branch=z9hG4bKPj119659756a7248239e0e53f69d49dc28
Max-Forwards: 70
From: “user6” sip:user6@AA.AAA.AAA.201;tag=c9ac3b5328ef4d1b9e5ed228ddf22a3f
To: “user6” sip:user6@AA.AAA.AAA.201
Contact: sip:52708419@SS.SS.SSS.254:53938;+sip.instance="urn:uuid:b878675c-66c9-493a-a3b4-5969bf6628e0"
Call-ID: 3512d0ea7680431396b42d03731428ef
CSeq: 1 REGISTER
Expires: 600
Supported: gruu
User-Agent: Blink 0.9.0 (Windows)
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to SS.SS.SSS.254:53938 (NAT)
Sending to SS.SS.SSS.254:53938 (NAT)

<— Transmitting (NAT) to SS.SS.SSS.254:53938 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;branch=z9hG4bKPj119659756a7248239e0e53f69d49dc28;received=SS.SS.SSS.254;rport=53938
From: “user6” sip:user6@AA.AAA.AAA.201;tag=c9ac3b5328ef4d1b9e5ed228ddf22a3f
To: “user6” sip:user6@AA.AAA.AAA.201;tag=as439c6143
Call-ID: 3512d0ea7680431396b42d03731428ef
CSeq: 1 REGISTER
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3e54f9dc"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3512d0ea7680431396b42d03731428ef’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:SS.SS.SSS.254:53938 —>
PUBLISH sip:user6@AA.AAA.AAA.201 SIP/2.0
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;rport;branch=z9hG4bKPjc0fea0fabb134e6fbe56bddbc1804f41
Max-Forwards: 70
From: “user6” sip:user6@AA.AAA.AAA.201;tag=2e78e2a8a4bc4ce4a7d2fd3a898a1889
To: “user6” sip:user6@AA.AAA.AAA.201
Call-ID: 1f0b6512ef544efca49ee4bb5afb3781
CSeq: 1 PUBLISH
Event: presence
Expires: 600
User-Agent: Blink 0.9.0 (Windows)
Content-Type: application/pidf+xml
Content-Length: 1824

<?xml version='1.0' encoding='UTF-8'?>

openagp-pidf:extendedavailable</agp-pidf:extended>caps:servcaps caps:audiotrue</caps:audio>caps:messagetrue</caps:message>caps:textfalse</caps:text>agp-caps:file-transfertrue</agp-caps:file-transfer>agp-caps:screen-sharing-servertrue</agp-caps:screen-sharing-server>agp-caps:screen-sharing-clienttrue</agp-caps:screen-sharing-client></caps:servcaps><c:display-name>user6</c:display-name><agp-pidf:device-info id=“b878675c-66c9-493a-a3b4-5969bf6628e0”>agp-pidf:descriptionwork7</agp-pidf:description>agp-pidf:user-agentBlink 0.9.0 (Windows)</agp-pidf:user-agent>agp-pidf:time-offset1200</agp-pidf:time-offset></agp-pidf:device-info><rpid:user-input idle-threshold=“600”>active</rpid:user-input>dm:deviceIDb878675c-66c9-493a-a3b4-5969bf6628e0</dm:deviceID>sip%3Auser6%40AA.AAA.AAA.2012014-08-11T08:41:49.185000-04:00<dm:person id=“PID-b5ad9106cbb9897534e06a0f446c836a”>rpid:activities rpid:otheravailable</rpid:other></rpid:activities>dm:timestamp2014-08-11T08:41:49.185000-04:00</dm:timestamp></dm:person><dm:device id=“DID-b878675c-66c9-493a-a3b4-5969bf6628e0”>dm:deviceIDb878675c-66c9-493a-a3b4-5969bf6628e0</dm:deviceID>dm:noteBlink 0.9.0 (Windows) at work7</dm:note>dm:timestamp2014-08-11T08:41:49.185000-04:00</dm:timestamp></dm:device>
<------------->
— (12 headers 2 lines) —
Sending to SS.SS.SSS.254:53938 (NAT)

<— Transmitting (NAT) to SS.SS.SSS.254:53938 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;branch=z9hG4bKPjc0fea0fabb134e6fbe56bddbc1804f41;received=SS.SS.SSS.254;rport=53938
From: “user6” sip:user6@AA.AAA.AAA.201;tag=2e78e2a8a4bc4ce4a7d2fd3a898a1889
To: “user6” sip:user6@AA.AAA.AAA.201;tag=as7f75fc21
Call-ID: 1f0b6512ef544efca49ee4bb5afb3781
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘1f0b6512ef544efca49ee4bb5afb3781’ Method: PUBLISH

<— SIP read from UDP:SS.SS.SSS.254:53938 —>
SUBSCRIBE sip:user6@AA.AAA.AAA.201 SIP/2.0
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;rport;branch=z9hG4bKPj5ee98ec745dc48c0bca26b739d8a4549
Max-Forwards: 70
From: “user6” sip:user6@AA.AAA.AAA.201;tag=8c7625cef79e4f56961eb3eaab106f6c
To: sip:user6@AA.AAA.AAA.201
Contact: sip:52708419@SS.SS.SSS.254:53938
Call-ID: 4c66dfd372734d39b1fcb8aebc9489db
CSeq: 18467 SUBSCRIBE
Event: message-summary
Expires: 600
Supported: 100rel, replaces, norefersub, gruu
Accept: application/simple-message-summary
Allow-Events: conference, message-summary, dialog, presence, presence.winfo, xcap-diff, dialog.winfo, refer
User-Agent: Blink 0.9.0 (Windows)
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to SS.SS.SSS.254:53938 (NAT)
Creating new subscription
Sending to SS.SS.SSS.254:53938 (NAT)
list_route: hop: sip:52708419@SS.SS.SSS.254:53938
Found peer ‘user6’ for ‘user6’ from SS.SS.SSS.254:53938

<— Transmitting (NAT) to SS.SS.SSS.254:53938 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;branch=z9hG4bKPj5ee98ec745dc48c0bca26b739d8a4549;received=SS.SS.SSS.254;rport=53938
From: “user6” sip:user6@AA.AAA.AAA.201;tag=8c7625cef79e4f56961eb3eaab106f6c
To: sip:user6@AA.AAA.AAA.201;tag=as77ffe622
Call-ID: 4c66dfd372734d39b1fcb8aebc9489db
CSeq: 18467 SUBSCRIBE
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[Aug 11 08:41:44] NOTICE[40118]: chan_sip.c:27928 handle_request_subscribe: Received SIP subscribe for peer without mailbox: user6
Really destroying SIP dialog ‘4c66dfd372734d39b1fcb8aebc9489db’ Method: SUBSCRIBE

<— SIP read from UDP:SS.SS.SSS.254:53938 —>
REGISTER sip:AA.AAA.AAA.201 SIP/2.0
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;rport;branch=z9hG4bKPj723b14f30ad84ea8aa9ef8f6fb7c0bd0
Max-Forwards: 70
From: “user6” sip:user6@AA.AAA.AAA.201;tag=c9ac3b5328ef4d1b9e5ed228ddf22a3f
To: “user6” sip:user6@AA.AAA.AAA.201
Contact: sip:52708419@SS.SS.SSS.254:53938;+sip.instance="urn:uuid:b878675c-66c9-493a-a3b4-5969bf6628e0"
Call-ID: 3512d0ea7680431396b42d03731428ef
CSeq: 2 REGISTER
Expires: 600
Supported: gruu
User-Agent: Blink 0.9.0 (Windows)
Authorization: Digest username=“user6”, realm=“asterisk”, nonce=“3e54f9dc”, uri=“sip:AA.AAA.AAA.201”, response=“99b0b3887b3f1ea53417556bd1531ad0”, algorithm=MD5
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to SS.SS.SSS.254:53938 (NAT)
Reliably Transmitting (NAT) to SS.SS.SSS.254:53938:
OPTIONS sip:52708419@SS.SS.SSS.254:53938 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK6a4465ba;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.126;tag=as2951df53
To: sip:52708419@SS.SS.SSS.254:53938
Contact: sip:asterisk@192.168.1.126:5060
Call-ID: 07c4773a513f9b160df270ac48880a3f@192.168.1.126:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.9.0
Date: Mon, 11 Aug 2014 12:41:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— Transmitting (NAT) to SS.SS.SSS.254:53938 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;branch=z9hG4bKPj723b14f30ad84ea8aa9ef8f6fb7c0bd0;received=SS.SS.SSS.254;rport=53938
From: “user6” sip:user6@AA.AAA.AAA.201;tag=c9ac3b5328ef4d1b9e5ed228ddf22a3f
To: “user6” sip:user6@AA.AAA.AAA.201;tag=as439c6143
Call-ID: 3512d0ea7680431396b42d03731428ef
CSeq: 2 REGISTER
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: sip:52708419@SS.SS.SSS.254:53938;expires=600
Date: Mon, 11 Aug 2014 12:41:44 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3512d0ea7680431396b42d03731428ef’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:SS.SS.SSS.254:53938 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.126:5060;rport=5060;received=AA.AAA.AAA.201;branch=z9hG4bK6a4465ba
Call-ID: 07c4773a513f9b160df270ac48880a3f@192.168.1.126:5060
From: “asterisk” sip:asterisk@192.168.1.126;tag=as2951df53
To: sip:52708419@192.168.7.132;tag=z9hG4bK6a4465ba
CSeq: 102 OPTIONS
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Accept: application/sdp, application/conference-info+xml, application/simple-message-summary, multipart/related, application/rlmi+xml, application/dialog-info+xml, multipart/related, application/rlmi+xml, application/pidf+xml, application/watcherinfo+xml, application/xcap-diff+xml, application/watcherinfo+xml, message/sipfrag;version=2.0
Supported: 100rel, replaces, norefersub, gruu
Server: Blink 0.9.0 (Windows)
Content-Length: 0

<------------->
— (11 headers 0 lines) —
[Aug 11 08:41:44] NOTICE[40118]: chan_sip.c:23645 handle_response_peerpoke: Peer ‘user6’ is now Reachable. (43ms / 2000ms)
Really destroying SIP dialog ‘07c4773a513f9b160df270ac48880a3f@192.168.1.126:5060’ Method: OPTIONS
Really destroying SIP dialog ‘fb224c2-71f1bbd6-9ce6ac6@III.III.II.131’ Method: OPTIONS

<— SIP read from UDP:SS.SS.SSS.254:53938 —>
PUBLISH sip:user6@AA.AAA.AAA.201 SIP/2.0
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;rport;branch=z9hG4bKPj641b6e20c5ee4942bf9a5635b49e15d5
Max-Forwards: 70
From: “user6” sip:user6@AA.AAA.AAA.201;tag=169f46dd6404442399d801ca1e22e405
To: “user6” sip:user6@AA.AAA.AAA.201
Call-ID: d6f786632efb4c378cfee3659ba4f5b1
CSeq: 1 PUBLISH
Event: presence
Expires: 600
User-Agent: Blink 0.9.0 (Windows)
Content-Type: application/pidf+xml
Content-Length: 1822

<?xml version='1.0' encoding='UTF-8'?>

openagp-pidf:extendedbusy</agp-pidf:extended>caps:servcaps caps:audiotrue</caps:audio>caps:messagetrue</caps:message>caps:textfalse</caps:text>agp-caps:file-transfertrue</agp-caps:file-transfer>agp-caps:screen-sharing-servertrue</agp-caps:screen-sharing-server>agp-caps:screen-sharing-clienttrue</agp-caps:screen-sharing-client></caps:servcaps><c:display-name>user6</c:display-name><agp-pidf:device-info id=“b878675c-66c9-493a-a3b4-5969bf6628e0”>agp-pidf:descriptionwork7</agp-pidf:description>agp-pidf:user-agentBlink 0.9.0 (Windows)</agp-pidf:user-agent>agp-pidf:time-offset1200</agp-pidf:time-offset></agp-pidf:device-info><rpid:user-input idle-threshold=“600”>active</rpid:user-input>dm:deviceIDb878675c-66c9-493a-a3b4-5969bf6628e0</dm:deviceID>sip%3Auser6%40AA.AAA.AAA.201On the phone2014-08-11T08:42:04.727000-04:00<dm:person id=“PID-b5ad9106cbb9897534e06a0f446c836a”>rpid:activities rpid:busy/</rpid:activities>dm:timestamp2014-08-11T08:42:04.727000-04:00</dm:timestamp></dm:person><dm:device id=“DID-b878675c-66c9-493a-a3b4-5969bf6628e0”>dm:deviceIDb878675c-66c9-493a-a3b4-5969bf6628e0</dm:deviceID>dm:noteBlink 0.9.0 (Windows) at work7</dm:note>dm:timestamp2014-08-11T08:42:04.727000-04:00</dm:timestamp></dm:device>
<------------->
— (12 headers 2 lines) —
Sending to SS.SS.SSS.254:53938 (NAT)

<— Transmitting (NAT) to SS.SS.SSS.254:53938 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;branch=z9hG4bKPj641b6e20c5ee4942bf9a5635b49e15d5;received=SS.SS.SSS.254;rport=53938
From: “user6” sip:user6@AA.AAA.AAA.201;tag=169f46dd6404442399d801ca1e22e405
To: “user6” sip:user6@AA.AAA.AAA.201;tag=as249e35a9
Call-ID: d6f786632efb4c378cfee3659ba4f5b1
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘d6f786632efb4c378cfee3659ba4f5b1’ Method: PUBLISH

<— SIP read from UDP:SS.SS.SSS.254:53938 —>
INVITE sip:user5@192.168.1.126 SIP/2.0
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;rport;branch=z9hG4bKPjf302d52b1f98454e8b77fd052cd03e41
Max-Forwards: 70
From: “user6” sip:user6@AA.AAA.AAA.201;tag=e65846a3cda84f6bad84d02b0c84bef2
To: sip:user5@192.168.1.126
Contact: sip:52708419@SS.SS.SSS.254:53938
Call-ID: c6c339def35c469e8c26cc1c9bc07a2d
CSeq: 8649 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, gruu
User-Agent: Blink 0.9.0 (Windows)
Content-Type: application/sdp
Content-Length: 510

v=0
o=- 3616735324 3616735324 IN IP4 192.168.7.132
s=Blink 0.9.0 (Windows)
t=0 0
m=audio 53946 RTP/AVP 0 8 113 9 101
c=IN IP4 SS.SS.SSS.254
a=rtcp:50001
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:113 opus/48000
a=fmtp:113 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y71cYd6MyNqA9ywtm0rS2KBRDByG6/7vuQnyeL8T
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:1H5vBzn2MCl6YbnXbD7VsZX2pyViiApknZOHtQmd
a=sendrecv
<------------->
— (13 headers 17 lines) —
Sending to SS.SS.SSS.254:53938 (NAT)
Sending to SS.SS.SSS.254:53938 (NAT)
Using INVITE request as basis request - c6c339def35c469e8c26cc1c9bc07a2d
Found peer ‘user6’ for ‘user6’ from SS.SS.SSS.254:53938
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 113
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format opus for ID 113
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
[Aug 11 08:42:00] ERROR[40118][C-0000000a]: chan_sip.c:33296 setup_srtp: No SRTP module loaded, can’t setup SRTP session.
[Aug 11 08:42:00] ERROR[40118][C-0000000a]: chan_sip.c:33296 setup_srtp: No SRTP module loaded, can’t setup SRTP session.
Capabilities: us - (ulaw|alaw|speex|h263p), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port SS.SS.SSS.254:53946
Looking for user5 in default (domain 192.168.1.126)
list_route: hop: sip:52708419@SS.SS.SSS.254:53938

<— Transmitting (NAT) to SS.SS.SSS.254:53938 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;branch=z9hG4bKPjf302d52b1f98454e8b77fd052cd03e41;received=SS.SS.SSS.254;rport=53938
From: “user6” sip:user6@AA.AAA.AAA.201;tag=e65846a3cda84f6bad84d02b0c84bef2
To: sip:user5@192.168.1.126
Call-ID: c6c339def35c469e8c26cc1c9bc07a2d
CSeq: 8649 INVITE
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:user5@192.168.1.126:5060
Content-Length: 0

<------------>
Audio is at 10438
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100009 (speex) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to CCC.CCC.CCC.59:33869:
INVITE sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK3f23d9ed;rport
Max-Forwards: 70
From: “user6” sip:user6@192.168.1.126;tag=as1afdf247
To: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP
Contact: sip:user6@192.168.1.126:5060
Call-ID: 1729ffb7568fc877705392010bfc93ec@192.168.1.126:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.9.0
Date: Mon, 11 Aug 2014 12:42:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1724485832 1724485832 IN IP4 192.168.1.126
s=Asterisk PBX 11.9.0
c=IN IP4 192.168.1.126
t=0 0
m=audio 10438 RTP/AVP 8 0 110 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #1 (NAT) to CCC.CCC.CCC.59:33869:
INVITE sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK3f23d9ed;rport
Max-Forwards: 70
From: “user6” sip:user6@192.168.1.126;tag=as1afdf247
To: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP
Contact: sip:user6@192.168.1.126:5060
Call-ID: 1729ffb7568fc877705392010bfc93ec@192.168.1.126:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.9.0
Date: Mon, 11 Aug 2014 12:42:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1724485832 1724485832 IN IP4 192.168.1.126
s=Asterisk PBX 11.9.0
c=IN IP4 192.168.1.126
t=0 0
m=audio 10438 RTP/AVP 8 0 110 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:CCC.CCC.CCC.59:33869 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK3f23d9ed;rport=5060;received=AA.AAA.AAA.201
To: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP
From: “user6” sip:user6@192.168.1.126;tag=as1afdf247
Call-ID: 1729ffb7568fc877705392010bfc93ec@192.168.1.126:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:CCC.CCC.CCC.59:33869 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK3f23d9ed;rport=5060;received=AA.AAA.AAA.201
To: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP
From: “user6” sip:user6@192.168.1.126;tag=as1afdf247
Call-ID: 1729ffb7568fc877705392010bfc93ec@192.168.1.126:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:CCC.CCC.CCC.59:33869 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK3f23d9ed;rport=5060;received=AA.AAA.AAA.201
Contact: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP
To: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP;tag=0a369b52
From: "user6"sip:user6@192.168.1.126;tag=as1afdf247
Call-ID: 1729ffb7568fc877705392010bfc93ec@192.168.1.126:5060
CSeq: 102 INVITE
User-Agent: Zoiper r26025
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP

<— Transmitting (NAT) to SS.SS.SSS.254:53938 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;branch=z9hG4bKPjf302d52b1f98454e8b77fd052cd03e41;received=SS.SS.SSS.254;rport=53938
From: “user6” sip:user6@AA.AAA.AAA.201;tag=e65846a3cda84f6bad84d02b0c84bef2
To: sip:user5@192.168.1.126;tag=as4f450ed2
Call-ID: c6c339def35c469e8c26cc1c9bc07a2d
CSeq: 8649 INVITE
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:user5@192.168.1.126:5060
Content-Length: 0

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

<— SIP read from UDP:CCC.CCC.CCC.59:33869 —>

<------------->
Really destroying SIP dialog ‘ZjlmYWQ4ZjE0YmRlZWE4MTBhNDg1MmE2ZjhkMjdlMWY.’ Method: REGISTER

<— SIP read from UDP:CCC.CCC.CCC.59:33869 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK3f23d9ed;rport=5060;received=AA.AAA.AAA.201
Contact: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP
To: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP;tag=0a369b52
From: "user6"sip:user6@192.168.1.126;tag=as1afdf247
Call-ID: 1729ffb7568fc877705392010bfc93ec@192.168.1.126:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r26025
Allow-Events: presence, kpml
Content-Length: 261

v=0
o=Zoiper 0 2 IN IP4 10.122.63.126
s=Zoiper
c=IN IP4 10.122.63.126
t=0 0
m=audio 52906 RTP/AVP 8 110 3 0 97
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=sendrecv
<------------->
— (13 headers 13 lines) —
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found audio description format speex for ID 110
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format iLBC for ID 97
Capabilities: us - (ulaw|alaw|speex|h263p), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|speex)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.122.63.126:52906
list_route: hop: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP
set_destination: Parsing sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP for address/port to send to
set_destination: set destination to 10.122.63.126:65391
Transmitting (NAT) to CCC.CCC.CCC.59:33869:
ACK sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK04a9b780;rport
Max-Forwards: 70
From: “user6” sip:user6@192.168.1.126;tag=as1afdf247
To: sip:user5@10.122.63.126:65391;rinstance=0e8a14616b0f04a1;transport=UDP;tag=0a369b52
Contact: sip:user6@192.168.1.126:5060
Call-ID: 1729ffb7568fc877705392010bfc93ec@192.168.1.126:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.9.0
Content-Length: 0

Audio is at 10392
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to SS.SS.SSS.254:53938 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP SS.SS.SSS.254:53938;branch=z9hG4bKPjf302d52b1f98454e8b77fd052cd03e41;received=SS.SS.SSS.254;rport=53938
From: “user6” sip:user6@AA.AAA.AAA.201;tag=e65846a3cda84f6bad84d02b0c84bef2
To: sip:user5@192.168.1.126;tag=as4f450ed2
Call-ID: c6c339def35c469e8c26cc1c9bc07a2d
CSeq: 8649 INVITE
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:user5@192.168.1.126:5060
Content-Type: application/sdp
Content-Length: 259