Jump to content

User Disconnects Call

Tony Yu

By Tony Yu
in General Setup

 Share

Recommended Posts

Tony Yu Newbie

Tony Yu

Posted
Posted

Hi there,

 

We are having intermittent issues where we drop calls. Sometimes after a dropped call I get an email saying User disconnects call in the subject. Email below. Also it seems that we cannot get past the 11 minute mark while on the phone. The call gets disconnected and the phone has to be rebooted in order to make more calls.

 

 

One side of the call between sip:15097207844@localhost;user=phone and sip:402@localhost did not receive media for 15.6 s and the other side of the call disconnected the call. The address of the other side was 66.23.129.253 (User-Agent=ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036). You may use this email as hint for a potential problem. The SIP messages are attached.

 

 

 

2012/2/27 14:50:29 Tx: tcp:192.168.1.10:53201 (281 bytes)

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-eoaf8rzrzc70;rport=53201

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 1 INVITE

Content-Length: 0

 

2012/2/27 14:50:29 Tx: tcp:192.168.1.10:53201 (495 bytes)

SIP/2.0 401 Authentication Required

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-eoaf8rzrzc70;rport=53201

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 1 INVITE

User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids

WWW-Authenticate: Digest realm="localhost",nonce="49a2d4d203befa6ee2e15b672eed4617",domain="sip:15097207844@localhost;user=phone",algorithm=MD5

Content-Length: 0

 

2012/2/27 14:50:30 Rx: tcp:192.168.1.10:53201 (420 bytes)

ACK sip:15097207844@localhost;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-eoaf8rzrzc70;rport

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:402@192.168.1.10:53201;transport=tcp;line=uuebwkg5>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

 

2012/2/27 14:50:30 Rx: tcp:192.168.1.10:53201 (1742 bytes)

INVITE sip:15097207844@localhost;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fslucigg78pa;rport

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 2 INVITE

Max-Forwards: 70

Contact: <sip:402@192.168.1.10:53201;transport=tcp;line=uuebwkg5>;reg-id=1

X-Serialnumber: 00041371061C

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom760/8.7.2.9

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Authorization: Digest username="402",realm="localhost",nonce="49a2d4d203befa6ee2e15b672eed4617",uri="sip:15097207844@localhost;user=phone",response="aaa9202c6872ccdb974a356b26449e79",algorithm=MD5

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 733

 

v=0

o=root 118993245 118993245 IN IP4 192.168.1.10

s=call

c=IN IP4 192.168.1.10

t=0 0

m=audio 59760 RTP/AVP 9 0 8 3 97 98 99 100 106 107 108 109 18 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8agvdPvzMo+D6DHUsQ9dT9t/rYStIyTEPtrHpab4

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 G726-16/8000

a=rtpmap:98 G726-24/8000

a=rtpmap:99 G726-32/8000

a=rtpmap:100 G726-40/8000

a=rtpmap:106 AAL2-G726-16/8000

a=rtpmap:107 AAL2-G726-24/8000

a=rtpmap:108 AAL2-G726-32/8000

a=rtpmap:109 AAL2-G726-40/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt

a=sendrecv

2012/2/27 14:50:30 Tx: tcp:192.168.1.10:53201 (281 bytes)

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fslucigg78pa;rport=53201

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 2 INVITE

Content-Length: 0

 

2012/2/27 14:50:30 Tx: tcp:192.168.1.10:53201 (900 bytes)

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fslucigg78pa;rport=53201

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 2 INVITE

Contact: <sip:402@192.168.1.26:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 298

 

v=0

o=- 1271747016 1271747016 IN IP4 192.168.1.26

s=-

c=IN IP4 192.168.1.26

t=0 0

m=audio 56720 RTP/AVP 0 18 101

a=rtpmap:0 pcmu/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

2012/2/27 14:50:30 Rx: tcp:192.168.1.10:53201 (587 bytes)

PRACK sip:402@192.168.1.26:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-851e4fsca841;rport

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 3 PRACK

Max-Forwards: 70

Contact: <sip:402@192.168.1.10:53201;transport=tcp;line=uuebwkg5>;reg-id=1

RAck: 1 2 INVITE

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE

Allow-Events: talk, hold, refer, call-info

Proxy-Require: buttons

Content-Length: 0

 

2012/2/27 14:50:30 Tx: tcp:192.168.1.10:53201 (380 bytes)

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-851e4fsca841;rport=53201

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 3 PRACK

Contact: <sip:402@192.168.1.26:5060;transport=tcp>

User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids

Content-Length: 0

 

2012/2/27 14:50:42 Tx: tcp:192.168.1.10:53201 (860 bytes)

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fslucigg78pa;rport=53201

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 2 INVITE

Contact: <sip:402@192.168.1.26:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids

Content-Type: application/sdp

Content-Length: 298

 

v=0

o=- 1271747016 1271747016 IN IP4 192.168.1.26

s=-

c=IN IP4 192.168.1.26

t=0 0

m=audio 56720 RTP/AVP 0 18 101

a=rtpmap:0 pcmu/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

2012/2/27 14:50:42 Rx: tcp:192.168.1.10:53201 (423 bytes)

ACK sip:402@192.168.1.26:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-o52xku6si97r;rport

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 2 ACK

Max-Forwards: 70

Contact: <sip:402@192.168.1.10:53201;transport=tcp;line=uuebwkg5>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

 

2012/2/27 14:51:03 Rx: tcp:192.168.1.10:53201 (597 bytes)

BYE sip:402@192.168.1.26:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fsyo66d8nyme;rport

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 4 BYE

Max-Forwards: 70

Contact: <sip:402@192.168.1.10:53201;transport=tcp;line=uuebwkg5>;reg-id=1

User-Agent: snom760/8.7.2.9

RTP-RxStat: Total_Rx_Pkts=894,Rx-Pkts=891,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=1663,Tx_Pkts=1663,Remote_Tx_Pkts=1663

Proxy-Require: buttons

Content-Length: 0

 

2012/2/27 14:51:03 Tx: tcp:192.168.1.10:53201 (378 bytes)

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fsyo66d8nyme;rport=53201

From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq

To: <sip:15097207844@localhost;user=phone>;tag=4f001275af

Call-ID: 73de4b4f0797-xa67qm6rn81r

CSeq: 4 BYE

Contact: <sip:402@192.168.1.26:5060;transport=tcp>

User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids

Content-Length: 0

 

 

 

==============================

 

2012/2/27 14:50:30 Tx: udp:66.23.129.253:5060 (1050 bytes)

INVITE sip:5097207844@nexvortex.com;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-8331bd4cc3c22de864eacf37e1fa6244;rport

From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

To: <sip:5097207844@localhost;user=phone>

Call-ID: 83be7b59@pbx

CSeq: 31301 INVITE

Max-Forwards: 70

Contact: <sip:Pe8345WcBd@192.168.1.26:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids

P-Asserted-Identity: <sip:8775784777>

Privacy: id

P-Charging-Vector: icid-value=;icid-generated-at=192.168.1.26;orig-ioi=localhost

Content-Type: application/sdp

Content-Length: 284

 

v=0

o=- 986850866 986850866 IN IP4 192.168.1.26

s=-

c=IN IP4 192.168.1.26

t=0 0

m=audio 55236 RTP/AVP 18 0 101

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

2012/2/27 14:50:30 Rx: udp:66.23.129.253:5060 (480 bytes)

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-8331bd4cc3c22de864eacf37e1fa6244;rport=5060

From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

To: <sip:5097207844@localhost;user=phone>;tag=9c6a9fdfd4d16ebaa52f34c4c528cbe5.63ba

Call-ID: 83be7b59@pbx

CSeq: 31301 INVITE

Proxy-Authenticate: Digest realm="nexvortex.com", nonce="4f4bdfa54d8a9eae60d6a580a08a1cf71ac78015"

Server: nVSIP 12.02.01

Content-Length: 0

 

2012/2/27 14:50:30 Tx: udp:66.23.129.253:5060 (1276 bytes)

INVITE sip:5097207844@nexvortex.com;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-97eadd1d07a98b29848791af7b631541;rport

From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

To: <sip:5097207844@localhost;user=phone>

Call-ID: 83be7b59@pbx

CSeq: 31302 INVITE

Max-Forwards: 70

Contact: <sip:Pe8345WcBd@192.168.1.26:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids

P-Asserted-Identity: <sip:8775784777>

Privacy: id

P-Charging-Vector: icid-value=;icid-generated-at=192.168.1.26;orig-ioi=localhost

Proxy-Authorization: Digest realm="nexvortex.com",nonce="4f4bdfa54d8a9eae60d6a580a08a1cf71ac78015",response="dc5f48a93ac0c225710f334d4d7ff85e",username="Pe8345WcBd",uri="sip:5097207844@nexvortex.com;user=phone",algorithm=MD5

Content-Type: application/sdp

Content-Length: 284

 

v=0

o=- 986850866 986850866 IN IP4 192.168.1.26

s=-

c=IN IP4 192.168.1.26

t=0 0

m=audio 55236 RTP/AVP 18 0 101

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

2012/2/27 14:50:30 Rx: udp:66.23.129.253:5060 (347 bytes)

SIP/2.0 100 trying -- your call is important to us

Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-97eadd1d07a98b29848791af7b631541;rport=5060

From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

To: <sip:5097207844@localhost;user=phone>

Call-ID: 83be7b59@pbx

CSeq: 31302 INVITE

Server: nVSIP 12.02.01

Content-Length: 0

 

2012/2/27 14:50:33 Rx: udp:66.23.129.253:5060 (979 bytes)

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-97eadd1d07a98b29848791af7b631541;rport=5060

From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

To: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001

Call-ID: 83be7b59@pbx

CSeq: 31302 INVITE

Record-Route: <sip:206.80.67.68;lr=on;ftag=327783872;osp-o=t5713901616956271617_s66.23.129.253:5060_a1330372217_c4>

Record-Route: <sip:5097207844@66.23.129.253:5060;nat=yes;ftag=327783872;lr=on>

Contact: <sip:15097207844@65.98.234.158:5060;transport=udp>

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

Content-Type: application/sdp

User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036

Content-Length: 223

 

v=0

o=- 1016715875 1016715875 IN IP4 65.98.234.158

s=ENSResip

c=IN IP4 65.98.234.158

t=0 0

m=audio 54878 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

2012/2/27 14:50:42 Rx: udp:66.23.129.253:5060 (965 bytes)

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-97eadd1d07a98b29848791af7b631541;rport=5060

From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

To: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001

Call-ID: 83be7b59@pbx

CSeq: 31302 INVITE

Record-Route: <sip:206.80.67.68;lr=on;ftag=327783872;osp-o=t5713901616956271617_s66.23.129.253:5060_a1330372217_c4>

Record-Route: <sip:5097207844@66.23.129.253:5060;nat=yes;ftag=327783872;lr=on>

Contact: <sip:15097207844@65.98.234.158:5060;transport=udp>

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

Content-Type: application/sdp

User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036

Content-Length: 223

 

v=0

o=- 1016715875 1016715875 IN IP4 65.98.234.158

s=ENSResip

c=IN IP4 65.98.234.158

t=0 0

m=audio 54878 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

2012/2/27 14:50:42 Tx: udp:66.23.129.253:5060 (756 bytes)

ACK sip:15097207844@65.98.234.158:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-64b09f91d60aa72f0f47543dc52c2990;rport

Route: <sip:5097207844@66.23.129.253:5060;nat=yes;ftag=327783872;lr=on>

Route: <sip:206.80.67.68;lr=on;ftag=327783872;osp-o=t5713901616956271617_s66.23.129.253:5060_a1330372217_c4>

From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

To: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001

Call-ID: 83be7b59@pbx

CSeq: 31302 ACK

Max-Forwards: 70

Contact: <sip:Pe8345WcBd@192.168.1.26:5060;transport=udp>

P-Asserted-Identity: <sip:8775784777>

Privacy: id

P-Charging-Vector: icid-value=;icid-generated-at=192.168.1.26;orig-ioi=localhost

Content-Length: 0

 

2012/2/27 14:50:43 Rx: udp:66.23.129.253:5060 (1178 bytes)

INVITE sip:Pe8345WcBd@192.168.1.26:5060;transport=udp SIP/2.0

Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on>

Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001>

Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK6178.c92882f6.0

Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.0

Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK3uscdq30384gi35927g0.1

Max-Forwards: 16

Contact: <sip:15097207844@65.98.234.158:5060;transport=udp>

To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001

Call-ID: 83be7b59@pbx

CSeq: 311201 INVITE

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

Content-Type: application/sdp

Date: Mon, 27 Feb 2012 19:49:55 GMT

User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036

Privacy: id

Content-Length: 223

 

v=0

o=- 1016715875 1016715876 IN IP4 65.98.234.158

s=ENSResip

c=IN IP4 65.98.234.158

t=0 0

m=audio 54878 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

2012/2/27 14:50:43 Tx: udp:66.23.129.253:5060 (1153 bytes)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK6178.c92882f6.0

Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.0

Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK3uscdq30384gi35927g0.1

Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on>

Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001>

From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001

To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

Call-ID: 83be7b59@pbx

CSeq: 311201 INVITE

Contact: <sip:Pe8345WcBd@192.168.1.26:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids

Content-Type: application/sdp

Content-Length: 249

 

v=0

o=- 986850866 986850866 IN IP4 192.168.1.26

s=-

c=IN IP4 192.168.1.26

t=0 0

m=audio 55236 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

2012/2/27 14:50:43 Rx: udp:66.23.129.253:5060 (1178 bytes)

INVITE sip:Pe8345WcBd@192.168.1.26:5060;transport=udp SIP/2.0

Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on>

Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001>

Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK6178.c92882f6.0

Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.0

Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK3uscdq30384gi35927g0.1

Max-Forwards: 16

Contact: <sip:15097207844@65.98.234.158:5060;transport=udp>

To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001

Call-ID: 83be7b59@pbx

CSeq: 311201 INVITE

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

Content-Type: application/sdp

Date: Mon, 27 Feb 2012 19:49:55 GMT

User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036

Privacy: id

Content-Length: 223

 

v=0

o=- 1016715875 1016715876 IN IP4 65.98.234.158

s=ENSResip

c=IN IP4 65.98.234.158

t=0 0

m=audio 54878 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

2012/2/27 14:50:43 Tr: udp:66.23.129.253:5060 (1153 bytes)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK6178.c92882f6.0

Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.0

Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK3uscdq30384gi35927g0.1

Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on>

Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001>

From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001

To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

Call-ID: 83be7b59@pbx

CSeq: 311201 INVITE

Contact: <sip:Pe8345WcBd@192.168.1.26:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids

Content-Type: application/sdp

Content-Length: 249

 

v=0

o=- 986850866 986850866 IN IP4 192.168.1.26

s=-

c=IN IP4 192.168.1.26

t=0 0

m=audio 55236 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

2012/2/27 14:50:43 Rx: udp:66.23.129.253:5060 (705 bytes)

ACK sip:Pe8345WcBd@192.168.1.26:5060;transport=udp SIP/2.0

Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on>

Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001>

Via: SIP/2.0/UDP 66.23.129.253:5060;branch=0

Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.2

Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK896l8230d08hka59c200.1

Max-Forwards: 16

To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001

Call-ID: 83be7b59@pbx

CSeq: 311201 ACK

User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036

Content-Length: 0

 

2012/2/27 14:50:43 Rx: udp:66.23.129.253:5060 (705 bytes)

ACK sip:Pe8345WcBd@192.168.1.26:5060;transport=udp SIP/2.0

Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on>

Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001>

Via: SIP/2.0/UDP 66.23.129.253:5060;branch=0

Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.2

Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK896l8230d08hka59c200.1

Max-Forwards: 16

To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872

From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001

Call-ID: 83be7b59@pbx

CSeq: 311201 ACK

User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036

Content-Length: 0

Link to comment
Share on other sites
Vodia PBX Grand Master

Vodia PBX

Posted
Posted

We are having intermittent issues where we drop calls. Sometimes after a dropped call I get an email saying User disconnects call in the subject. Email below. Also it seems that we cannot get past the 11 minute mark while on the phone. The call gets disconnected and the phone has to be rebooted in order to make more calls.

 

That is obviously a problem of the phone. The 7xx series had a couple of software images over the past few months. Not sure which is the latest one, but you need a new image.

 

One side of the call between sip:15097207844@localhost;user=phone and sip:402@localhost did not receive media for 15.6 s and the other side of the call disconnected the call. The address of the other side was 66.23.129.253 (User-Agent=ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036). You may use this email as hint for a potential problem. The SIP messages are attached.

 

That is a sign that there is trouble with one-way audio on hte trunk side. When the user disconnects the call, the PBX checks if the phone had to chance to hear anything, and in this case that seems not to be the case. It seems that your service provider does support the PBX running on a unroutable IP address, otherwise you would never get the INVITE from the SP. At first glance, it seems there is trouble with the firewall. Is this a cable modem router?

Link to comment
Share on other sites
Tony Yu Newbie

Tony Yu

Posted
  • Author
Posted

That is obviously a problem of the phone. The 7xx series had a couple of software images over the past few months. Not sure which is the latest one, but you need a new image.

 

 

 

That is a sign that there is trouble with one-way audio on hte trunk side. When the user disconnects the call, the PBX checks if the phone had to chance to hear anything, and in this case that seems not to be the case. It seems that your service provider does support the PBX running on a unroutable IP address, otherwise you would never get the INVITE from the SP. At first glance, it seems there is trouble with the firewall. Is this a cable modem router?

 

 

Not sure what you mean by software images? Are we talking about firmware?

 

We have a T1 line running clean into PBX with no firewalls. Any ideas?

 

Thanks,

Link to comment
Share on other sites
pbx support Newbie

pbx support

Posted
Posted

Looking at the log, the call was established only for 21 seconds.

 

PBX sends 200 OK to the phone (760)

2012/2/27 14:50:42 Tx: tcp:192.168.1.10:53201 (860 bytes)
SIP/2.0 200 Ok

 

 

The user at 760 disconnects the call after not getting any media for 15.6 s.

2012/2/27 14:51:03 Rx: tcp:192.168.1.10:53201 (597 bytes)
BYE sip:402@192.168.1.26:5060;transport=tcp SIP/2.0

 

The issue could be anywhere all the way from the remote device/remote network/local network/Gateway/PBX/Phone etc

 

Just curious, is the remote side MagicJack?

Link to comment
Share on other sites
Tony Yu Newbie

Tony Yu

Posted
  • Author
Posted

Looking at the log, the call was established only for 21 seconds.

 

PBX sends 200 OK to the phone (760)

2012/2/27 14:50:42 Tx: tcp:192.168.1.10:53201 (860 bytes)
SIP/2.0 200 Ok

 

 

The user at 760 disconnects the call after not getting any media for 15.6 s.

2012/2/27 14:51:03 Rx: tcp:192.168.1.10:53201 (597 bytes)
BYE sip:402@192.168.1.26:5060;transport=tcp SIP/2.0

 

The issue could be anywhere all the way from the remote device/remote network/local network/Gateway/PBX/Phone etc

 

Just curious, is the remote side MagicJack?

 

Nope, the remote side is NexVortex in US

 

Any suggestions to help with my problem?

Link to comment
Share on other sites
Vodia PBX Grand Master

Vodia PBX

Posted
Posted

I would make sure that internal calls work fine. Make a couple of calls from the 760 to another phone and check if they are okay. If thats the case the problem is related to the SIP trunk. There we would have to focus on NAT related problems with the firewall, as they are most of the time the problem.

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

×
 Share
Go to topic listing


×
×
  • Create New...