Jump to content

Integration with Avaya system


Ganesh
 Share

Recommended Posts

We are integrating pbxnsip with Avaya system on SIP trunk. We have configured SIP trunk (gateway mode) on pbxnsip and Avaya. But unable to route calls both ways between the two system. Attached is the logs and wireshark traces captured.

 

4229 and 4431 is extension (Avaya phones) in Avaya. 2201 and 2202 is extensions (Snom phones) on pbxnsip. Avaya uses port 5061 and supports only TLS on SIP trunk.

 

Please let me know if you can get some information from these logs. Below is one more log taken from pbxnsip while trying to call Avaya phones.

 

[5] 2008/07/07 04:23:17: SIP port accept from 192.168.192.28:14935

[7] 2008/07/07 04:23:20: SIP Rx udp:192.168.136.36:2051:

REGISTER sip:192.168.192.50 SIP/2.0

Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-81zc3zm6vbe4;rport

From: <sip:5203@192.168.192.50>;tag=3eez6gof7x

To: <sip:5203@192.168.192.50>

Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7

CSeq: 1033 REGISTER

Max-Forwards: 70

Contact: <sip:5203@192.168.136.36:2051;line=vfooudzb>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:519aec13-5a5d-4f46-92d3-aa8451bb25aa>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"

User-Agent: snom300/6.5.13

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.136.36

WWW-Contact: <http://192.168.136.36:80>

WWW-Contact: <https://192.168.136.36:443>

Expires: 3600

Content-Length: 0

 

 

[9] 2008/07/07 04:23:20: Resolve 47: aaaa udp 192.168.136.36 2051

[9] 2008/07/07 04:23:20: Resolve 47: a udp 192.168.136.36 2051

[9] 2008/07/07 04:23:20: Resolve 47: udp 192.168.136.36 2051

[7] 2008/07/07 04:23:20: SIP Tx udp:192.168.136.36:2051:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-81zc3zm6vbe4;rport=2051

From: <sip:5203@192.168.192.50>;tag=3eez6gof7x

To: <sip:5203@192.168.192.50>;tag=1ee70c8e7e

Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7

CSeq: 1033 REGISTER

Content-Length: 0

 

 

[7] 2008/07/07 04:23:26: SIP Rx tls:192.168.25.103:2053:

INVITE sip:4229@192.168.192.50;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport

From: <sip:2201@192.168.192.50>;tag=p10csqj5hf

To: <sip:4229@192.168.192.50;user=phone>

Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1

P-Key-Flags: keys="3"

User-Agent: snom300/6.5.13

Accept: application/sdp

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

Allow-Events: talk, hold, refer

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Min-SE: 90

Content-Type: application/sdp

Content-Length: 345

 

v=0

o=root 576733664 576733664 IN IP4 192.168.25.103

s=call

c=IN IP4 192.168.25.103

t=0 0

m=audio 58152 RTP/AVP 18 4 0 8 3 9 2 101

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:9 g722/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[8] 2008/07/07 04:23:26: Packet authenticated by transport layer

[7] 2008/07/07 04:23:26: UDP: Opening socket on port 52908

[7] 2008/07/07 04:23:26: UDP: Opening socket on port 52909

[8] 2008/07/07 04:23:26: Could not find a trunk (1 trunks)

[9] 2008/07/07 04:23:26: Using outbound proxy sip:192.168.25.103:2053;transport=tls because of flow-label

[9] 2008/07/07 04:23:26: Resolve 48: tls 192.168.25.103 2053

[7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport=2053

From: <sip:2201@192.168.192.50>;tag=p10csqj5hf

To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48

Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE

CSeq: 1 INVITE

Content-Length: 0

 

 

[7] 2008/07/07 04:23:26: Set packet length to 20

[6] 2008/07/07 04:23:26: Sending RTP for 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE#1c9ec29c48 to 192.168.25.103:58152

[9] 2008/07/07 04:23:26: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 4229@192.168.192.50

[5] 2008/07/07 04:23:26: Dialplan New: Match 4229@192.168.192.50 to <sip:4229@192.168.192.28:5061;user=phone> on trunk SIP

[8] 2008/07/07 04:23:26: Play audio_moh/noise.wav

[7] 2008/07/07 04:23:26: UDP: Opening socket on port 49714

[7] 2008/07/07 04:23:26: UDP: Opening socket on port 49715

[9] 2008/07/07 04:23:26: Resolve 49: url sip:192.168.192.28:5061

[9] 2008/07/07 04:23:26: Resolve 49: udp 192.168.192.28 5061

[7] 2008/07/07 04:23:26: SIP Tx udp:192.168.192.28:5061:

INVITE sip:4229@192.168.192.28:5061;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.192.50:5060;branch=z9hG4bK-017e34d40401d0870149413127470191;rport

From: <sip:2201@localhost>;tag=48313

To: <sip:4229@192.168.192.28:5061;user=phone>

Call-ID: 910d81bb@pbx

CSeq: 7463 INVITE

Max-Forwards: 70

Contact: <sip:2201@192.168.192.50:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Type: application/sdp

Content-Length: 294

 

v=0

o=- 56787 56787 IN IP4 192.168.192.50

s=-

c=IN IP4 192.168.192.50

t=0 0

m=audio 49714 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/07/07 04:23:26: Set packet length to 20

[9] 2008/07/07 04:23:26: Resolve 50: tls 192.168.25.103 2053

[7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053:

SIP/2.0 183 Ringing

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport=2053

From: <sip:2201@192.168.192.50>;tag=p10csqj5hf

To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48

Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE

CSeq: 1 INVITE

Contact: <sip:2201@192.168.192.50:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 304

 

v=0

o=- 7292 7292 IN IP4 192.168.192.50

s=-

c=IN IP4 192.168.192.50

t=0 0

m=audio 52908 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[8] 2008/07/07 04:23:26: UDP: recvfrom receives ICMP message

[5] 2008/07/07 04:23:26: Connection refused on udp:192.168.192.28:5061

[6] 2008/07/07 04:23:26: Could not determine destination address on 49

[7] 2008/07/07 04:23:26: Call 910d81bb@pbx#48313: Clear last INVITE

[9] 2008/07/07 04:23:26: Resolve 51: tls 192.168.25.103 2053

[7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053:

SIP/2.0 500 Network Failure

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport=2053

From: <sip:2201@192.168.192.50>;tag=p10csqj5hf

To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48

Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE

CSeq: 1 INVITE

Contact: <sip:2201@192.168.192.50:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Length: 0

 

 

[7] 2008/07/07 04:23:26: SIP Rx tls:192.168.25.103:2053:

PRACK sip:2201@192.168.192.50:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-on4zow98h369;rport

From: <sip:2201@192.168.192.50>;tag=p10csqj5hf

To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48

Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE

CSeq: 2 PRACK

Max-Forwards: 70

Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1

RAck: 1 1 INVITE

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

Allow-Events: talk, hold, refer

Content-Length: 0

 

 

[8] 2008/07/07 04:23:26: Packet authenticated by transport layer

[9] 2008/07/07 04:23:26: Resolve 52: tls 192.168.25.103 2053

[7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-on4zow98h369;rport=2053

From: <sip:2201@192.168.192.50>;tag=p10csqj5hf

To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48

Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE

CSeq: 2 PRACK

Contact: <sip:2201@192.168.192.50:5061;transport=tls>

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Length: 0

 

 

[7] 2008/07/07 04:23:26: SIP Rx tls:192.168.25.103:2053:

ACK sip:4229@192.168.192.50;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport

From: <sip:2201@192.168.192.50>;tag=p10csqj5hf

To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48

Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1

Content-Length: 0

 

 

[8] 2008/07/07 04:23:26: Packet authenticated by transport layer

[7] 2008/07/07 04:23:26: Other Ports: 1

[7] 2008/07/07 04:23:26: Call Port: 910d81bb@pbx#48313

[7] 2008/07/07 04:23:30: SIP Rx tls:192.168.25.103:2053:

INVITE sip:4431@192.168.192.50;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport

From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw

To: <sip:4431@192.168.192.50;user=phone>

Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1

P-Key-Flags: keys="3"

User-Agent: snom300/6.5.13

Accept: application/sdp

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

Allow-Events: talk, hold, refer

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Min-SE: 90

Content-Type: application/sdp

Content-Length: 347

 

v=0

o=root 1459444772 1459444772 IN IP4 192.168.25.103

s=call

c=IN IP4 192.168.25.103

t=0 0

m=audio 58646 RTP/AVP 18 4 0 8 3 9 2 101

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:9 g722/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[8] 2008/07/07 04:23:30: Packet authenticated by transport layer

[7] 2008/07/07 04:23:30: UDP: Opening socket on port 52682

[7] 2008/07/07 04:23:30: UDP: Opening socket on port 52683

[8] 2008/07/07 04:23:30: Could not find a trunk (1 trunks)

[9] 2008/07/07 04:23:30: Using outbound proxy sip:192.168.25.103:2053;transport=tls because of flow-label

[9] 2008/07/07 04:23:30: Resolve 53: tls 192.168.25.103 2053

[7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport=2053

From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw

To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863

Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE

CSeq: 1 INVITE

Content-Length: 0

 

 

[7] 2008/07/07 04:23:30: Set packet length to 20

[6] 2008/07/07 04:23:30: Sending RTP for 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE#6f162b9863 to 192.168.25.103:58646

[9] 2008/07/07 04:23:30: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 4431@192.168.192.50

[5] 2008/07/07 04:23:30: Dialplan New: Match 4431@192.168.192.50 to <sip:4431@192.168.192.28:5061;user=phone> on trunk SIP

[8] 2008/07/07 04:23:30: Play audio_moh/noise.wav

[7] 2008/07/07 04:23:30: UDP: Opening socket on port 52286

[7] 2008/07/07 04:23:30: UDP: Opening socket on port 52287

[9] 2008/07/07 04:23:30: Resolve 54: url sip:192.168.192.28:5061

[9] 2008/07/07 04:23:30: Resolve 54: udp 192.168.192.28 5061

[7] 2008/07/07 04:23:30: SIP Tx udp:192.168.192.28:5061:

INVITE sip:4431@192.168.192.28:5061;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.192.50:5060;branch=z9hG4bK-f969c5b8969691bf078c04d44f93e63f;rport

From: <sip:2201@localhost>;tag=17880

To: <sip:4431@192.168.192.28:5061;user=phone>

Call-ID: 63864075@pbx

CSeq: 29415 INVITE

Max-Forwards: 70

Contact: <sip:2201@192.168.192.50:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Type: application/sdp

Content-Length: 292

 

v=0

o=- 2970 2970 IN IP4 192.168.192.50

s=-

c=IN IP4 192.168.192.50

t=0 0

m=audio 52286 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/07/07 04:23:30: Set packet length to 20

[9] 2008/07/07 04:23:30: Resolve 55: tls 192.168.25.103 2053

[7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053:

SIP/2.0 183 Ringing

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport=2053

From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw

To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863

Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE

CSeq: 1 INVITE

Contact: <sip:2201@192.168.192.50:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 306

 

v=0

o=- 28977 28977 IN IP4 192.168.192.50

s=-

c=IN IP4 192.168.192.50

t=0 0

m=audio 52682 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[8] 2008/07/07 04:23:30: UDP: recvfrom receives ICMP message

[5] 2008/07/07 04:23:30: Connection refused on udp:192.168.192.28:5061

[6] 2008/07/07 04:23:30: Could not determine destination address on 54

[7] 2008/07/07 04:23:30: Call 63864075@pbx#17880: Clear last INVITE

[9] 2008/07/07 04:23:30: Resolve 56: tls 192.168.25.103 2053

[7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053:

SIP/2.0 500 Network Failure

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport=2053

From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw

To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863

Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE

CSeq: 1 INVITE

Contact: <sip:2201@192.168.192.50:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Length: 0

 

 

[7] 2008/07/07 04:23:30: SIP Rx tls:192.168.25.103:2053:

PRACK sip:2201@192.168.192.50:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-xb9k7e6khimn;rport

From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw

To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863

Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE

CSeq: 2 PRACK

Max-Forwards: 70

Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1

RAck: 1 1 INVITE

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

Allow-Events: talk, hold, refer

Content-Length: 0

 

 

[8] 2008/07/07 04:23:30: Packet authenticated by transport layer

[9] 2008/07/07 04:23:30: Resolve 57: tls 192.168.25.103 2053

[7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-xb9k7e6khimn;rport=2053

From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw

To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863

Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE

CSeq: 2 PRACK

Contact: <sip:2201@192.168.192.50:5061;transport=tls>

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Length: 0

 

 

[7] 2008/07/07 04:23:30: SIP Rx tls:192.168.25.103:2053:

ACK sip:4431@192.168.192.50;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport

From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw

To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863

Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1

Content-Length: 0

 

 

[8] 2008/07/07 04:23:30: Packet authenticated by transport layer

[7] 2008/07/07 04:23:30: Other Ports: 2

[7] 2008/07/07 04:23:30: Call Port: 63864075@pbx#17880

[7] 2008/07/07 04:23:30: Call Port: 910d81bb@pbx#48313

[8] 2008/07/07 04:23:34: Hangup: Call 910d81bb@pbx#48313 not found

[7] 2008/07/07 04:23:36: SIP Rx udp:192.168.136.36:2051:

REGISTER sip:192.168.192.50 SIP/2.0

Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-e5flcagayes3;rport

From: <sip:5203@192.168.192.50>;tag=cm2h1pdaec

To: <sip:5203@192.168.192.50>

Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7

CSeq: 1034 REGISTER

Max-Forwards: 70

Contact: <sip:5203@192.168.136.36:2051;line=vfooudzb>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:519aec13-5a5d-4f46-92d3-aa8451bb25aa>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"

User-Agent: snom300/6.5.13

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.136.36

WWW-Contact: <http://192.168.136.36:80>

WWW-Contact: <https://192.168.136.36:443>

Expires: 3600

Content-Length: 0

 

 

[9] 2008/07/07 04:23:36: Resolve 58: aaaa udp 192.168.136.36 2051

[9] 2008/07/07 04:23:36: Resolve 58: a udp 192.168.136.36 2051

[9] 2008/07/07 04:23:36: Resolve 58: udp 192.168.136.36 2051

[7] 2008/07/07 04:23:36: SIP Tx udp:192.168.136.36:2051:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-e5flcagayes3;rport=2051

From: <sip:5203@192.168.192.50>;tag=cm2h1pdaec

To: <sip:5203@192.168.192.50>;tag=1ee70c8e7e

Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7

CSeq: 1034 REGISTER

Content-Length: 0

 

 

[8] 2008/07/07 04:23:38: Hangup: Call 63864075@pbx#17880 not found

[7] 2008/07/07 04:23:45: SIP Rx tls:192.168.25.103:2053:

REGISTER sip:192.168.192.50 SIP/2.0

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-8ruwlxvmnsxh;rport

From: <sip:2201@192.168.192.50>;tag=zn5jibw3c8

To: <sip:2201@192.168.192.50>

Call-ID: 3c267013a604-k4l8r8hzrhkc@snom300-0004132889BE

CSeq: 5 REGISTER

Max-Forwards: 70

Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:5c780463-7a16-4199-bb5c-a029eae57121>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"

User-Agent: snom300/6.5.13

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.25.103

WWW-Contact: <http://192.168.25.103:80>

WWW-Contact: <https://192.168.25.103:443>

Expires: 3600

Content-Length: 0

 

 

[8] 2008/07/07 04:23:45: Packet authenticated by transport layer

[9] 2008/07/07 04:23:45: Resolve 59: tls 192.168.25.103 2053

[7] 2008/07/07 04:23:45: SIP Tx tls:192.168.25.103:2053:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-8ruwlxvmnsxh;rport=2053

From: <sip:2201@192.168.192.50>;tag=zn5jibw3c8

To: <sip:2201@192.168.192.50>;tag=f9ed8b9df9

Call-ID: 3c267013a604-k4l8r8hzrhkc@snom300-0004132889BE

CSeq: 5 REGISTER

Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;expires=178

Content-Length: 0

 

 

[5] 2008/07/07 04:23:46: SIP port accept from 192.168.192.28:14946

[7] 2008/07/07 04:23:51: SIP Rx udp:192.168.136.36:2051:

REGISTER sip:192.168.192.50 SIP/2.0

Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-amoqwm4o901i;rport

From: <sip:5203@192.168.192.50>;tag=g4q9ekm9mp

To: <sip:5203@192.168.192.50>

Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7

CSeq: 1035 REGISTER

Max-Forwards: 70

Contact: <sip:5203@192.168.136.36:2051;line=vfooudzb>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:519aec13-5a5d-4f46-92d3-aa8451bb25aa>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"

User-Agent: snom300/6.5.13

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.136.36

WWW-Contact: <http://192.168.136.36:80>

WWW-Contact: <https://192.168.136.36:443>

Expires: 3600

Content-Length: 0

 

 

[9] 2008/07/07 04:23:51: Resolve 60: aaaa udp 192.168.136.36 2051

[9] 2008/07/07 04:23:51: Resolve 60: a udp 192.168.136.36 2051

[9] 2008/07/07 04:23:51: Resolve 60: udp 192.168.136.36 2051

[7] 2008/07/07 04:23:51: SIP Tx udp:192.168.136.36:2051:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-amoqwm4o901i;rport=2051

From: <sip:5203@192.168.192.50>;tag=g4q9ekm9mp

To: <sip:5203@192.168.192.50>;tag=1ee70c8e7e

Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7

CSeq: 1035 REGISTER

Content-Length: 0

Wireshark_SIP_logs1.zip

SIP_logs1.txt

Link to comment
Share on other sites

  • 2 weeks later...

We configured the outbound proxy and specified to use TLS (as Avaya only supports TLS on direct SIP trunk without Avaya SES). We were still unable to route calls between the two systems. Below is the logs captured. 59999 is the Avaya phone extension.

 

This link (http://www.avayausers.com/showthread.php?t=10700) says that "TCP Sip is supported, but UDP SIP is not supported without the Avaya Sip Server. (SES) Even then you still need a session border controller" (we are not using the Avaya SES here).

It also says SBC is required while using the SIP trunk on Avaya.

 

Since we are integrating this in the same network (LAN), do we need a SBC? Also I think SBC is inbuilt in PBXNSIP. Right?

 

Should i try using a SBC or is there any other settings we need to do from our side. Below is the logs captured from PBXNSIP.

 

[7] 2008/07/16 06:45:20: SIP Rx udp:192.168.38.21:12163:

INVITE sip:59999@192.168.38.21 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:79999@192.168.38.21:12163>

To: "59999"<sip:59999@192.168.38.21>

From: "79999"<sip:79999@192.168.38.21>;tag=20694b71

Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.

CSeq: 1 INVITE

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

Content-Type: application/sdp

User-Agent: X-Lite release 1100l stamp 47546

Content-Length: 423

 

v=0

o=- 6 2 IN IP4 192.168.38.21

s=CounterPath X-Lite 3.0

c=IN IP4 192.168.38.21

t=0 0

m=audio 3622 RTP/AVP 107 119 100 106 0 105 98 8 3 101

a=alt:1 1 : qqzBZNsZ mIyNWmYl 192.168.38.21 3622

a=fmtp:101 0-15

a=rtpmap:107 BV32/16000

a=rtpmap:119 BV32-FEC/16000

a=rtpmap:100 SPEEX/16000

a=rtpmap:106 SPEEX-FEC/16000

a=rtpmap:105 SPEEX-FEC/8000

a=rtpmap:98 iLBC/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[7] 2008/07/16 06:45:20: UDP: Opening socket on port 49246

[7] 2008/07/16 06:45:20: UDP: Opening socket on port 49247

[8] 2008/07/16 06:45:20: Could not find a trunk (1 trunks)

[8] 2008/07/16 06:45:20: Using outbound proxy sip:192.168.38.21:12163;transport=udp because UDP packet source did not match the via header

[9] 2008/07/16 06:45:20: Resolve 50: udp 192.168.38.21 12163

[7] 2008/07/16 06:45:20: SIP Tx udp:192.168.38.21:12163:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21

From: "79999" <sip:79999@192.168.38.21>;tag=20694b71

To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412

Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.

CSeq: 1 INVITE

Content-Length: 0

 

 

[6] 2008/07/16 06:45:20: Sending RTP for MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.#6325fc9412 to 192.168.38.21:3622

[9] 2008/07/16 06:45:21: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 59999@192.168.38.21

[5] 2008/07/16 06:45:21: Dialplan New: Match 59999@192.168.38.21 to <sip:59999@192.168.38.20;user=phone> on trunk SIP

[8] 2008/07/16 06:45:21: Play audio_moh/noise.wav

[7] 2008/07/16 06:45:21: UDP: Opening socket on port 59432

[7] 2008/07/16 06:45:21: UDP: Opening socket on port 59433

[9] 2008/07/16 06:45:21: Resolve 51: url sip:192.168.38.20:5061;transport=tls

[9] 2008/07/16 06:45:21: Resolve 51: a tls 192.168.38.20 5061

[9] 2008/07/16 06:45:21: Resolve 51: tls 192.168.38.20 5061

[7] 2008/07/16 06:45:21: SIP Tx tls:192.168.38.20:5061:

INVITE sip:59999@192.168.38.20;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.38.21:1210;branch=z9hG4bK-48a3669a14150ffd1e6b3c48e9c5f659;rport

From: <sip:79999@localhost>;tag=5447

To: <sip:59999@192.168.38.20;user=phone>

Call-ID: ff5f3792@pbx

CSeq: 6271 INVITE

Max-Forwards: 70

Contact: <sip:79999@192.168.38.21:1210;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Type: application/sdp

Content-Length: 423

 

v=0

o=- 28504 28504 IN IP4 192.168.38.21

s=-

c=IN IP4 192.168.38.21

t=0 0

m=audio 59432 RTP/AVP 0 8 9 18 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:UxcnvE8+sfevfgeIrnn35dXnjcuAf3Ikos1Dnk3f

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[9] 2008/07/16 06:45:21: Resolve 52: udp 192.168.38.21 12163

[7] 2008/07/16 06:45:21: SIP Tx udp:192.168.38.21:12163:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21

From: "79999" <sip:79999@192.168.38.21>;tag=20694b71

To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412

Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.

CSeq: 1 INVITE

Contact: <sip:79999@127.0.0.1:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Type: application/sdp

Content-Length: 233

 

v=0

o=- 16981 16981 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 49246 RTP/AVP 0 8 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[6] 2008/07/16 06:45:21: Sending RTP for MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.#6325fc9412 to 127.0.0.1:3622

[7] 2008/07/16 06:45:21: SIP Tr udp:192.168.38.21:12163:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21

From: "79999" <sip:79999@192.168.38.21>;tag=20694b71

To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412

Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.

CSeq: 1 INVITE

Contact: <sip:79999@127.0.0.1:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Type: application/sdp

Content-Length: 233

 

v=0

o=- 16981 16981 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 49246 RTP/AVP 0 8 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/07/16 06:45:22: SIP Rx udp:192.168.38.21:12163:

REGISTER sip:192.168.38.21 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-8e7bc46eb755e302-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14>

To: "79999"<sip:79999@192.168.38.21>

From: "79999"<sip:79999@192.168.38.21>;tag=0948d273

Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ.

CSeq: 26 REGISTER

Expires: 3600

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

User-Agent: X-Lite release 1100l stamp 47546

Content-Length: 0

 

 

[9] 2008/07/16 06:45:22: Resolve 53: udp 192.168.38.21 12163

[7] 2008/07/16 06:45:22: SIP Tx udp:192.168.38.21:12163:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-8e7bc46eb755e302-1---d8754z-;rport=12163;received=192.168.38.21

From: "79999" <sip:79999@192.168.38.21>;tag=0948d273

To: "79999" <sip:79999@192.168.38.21>;tag=11399c85b8

Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ.

CSeq: 26 REGISTER

Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14>;expires=31

Content-Length: 0

 

 

[7] 2008/07/16 06:45:22: SIP Tr udp:192.168.38.21:12163:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21

From: "79999" <sip:79999@192.168.38.21>;tag=20694b71

To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412

Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.

CSeq: 1 INVITE

Contact: <sip:79999@127.0.0.1:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Type: application/sdp

Content-Length: 233

 

v=0

o=- 16981 16981 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 49246 RTP/AVP 0 8 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/07/16 06:45:41: Last message repeated 4 times

 

[5] 2008/07/16 06:45:41: SIP port accept from 192.168.38.14:24434

[7] 2008/07/16 06:45:48: SIP Rx udp:192.168.38.21:12163:

REGISTER sip:192.168.38.21 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-0e56b011be31d71c-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14>

To: "79999"<sip:79999@192.168.38.21>

From: "79999"<sip:79999@192.168.38.21>;tag=0948d273

Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ.

CSeq: 27 REGISTER

Expires: 3600

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

User-Agent: X-Lite release 1100l stamp 47546

Content-Length: 0

 

 

[9] 2008/07/16 06:45:48: Resolve 54: udp 192.168.38.21 12163

[7] 2008/07/16 06:45:48: SIP Tx udp:192.168.38.21:12163:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-0e56b011be31d71c-1---d8754z-;rport=12163;received=192.168.38.21

From: "79999" <sip:79999@192.168.38.21>;tag=0948d273

To: "79999" <sip:79999@192.168.38.21>;tag=11399c85b8

Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ.

CSeq: 27 REGISTER

Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14>;expires=29

Content-Length: 0

 

 

[7] 2008/07/16 06:45:51: Call ff5f3792@pbx#5447: Clear last INVITE

[9] 2008/07/16 06:45:51: Resolve 55: udp 192.168.38.21 12163

[7] 2008/07/16 06:45:51: SIP Tx udp:192.168.38.21:12163:

SIP/2.0 408 Request Timeout

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21

From: "79999" <sip:79999@192.168.38.21>;tag=20694b71

To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412

Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.

CSeq: 1 INVITE

Contact: <sip:79999@127.0.0.1:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.6.2450

Content-Length: 0

 

 

[7] 2008/07/16 06:45:51: SIP Rx udp:192.168.38.21:12163:

ACK sip:59999@192.168.38.21 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport

To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412

From: "79999"<sip:79999@192.168.38.21>;tag=20694b71

Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.

CSeq: 1 ACK

Content-Length: 0

 

 

[7] 2008/07/16 06:45:51: Other Ports: 1

[7] 2008/07/16 06:45:51: Call Port: ff5f3792@pbx#5447

[8] 2008/07/16 06:45:59: Hangup: Call ff5f3792@pbx#5447 not found

Link to comment
Share on other sites

We configured the outbound proxy and specified to use TLS (as Avaya only supports TLS on direct SIP trunk without Avaya SES). We were still unable to route calls between the two systems. Below is the logs captured. 59999 is the Avaya phone extension.

 

This link (http://www.avayausers.com/showthread.php?t=10700) says that "TCP Sip is supported, but UDP SIP is not supported without the Avaya Sip Server. (SES) Even then you still need a session border controller" (we are not using the Avaya SES here).

It also says SBC is required while using the SIP trunk on Avaya.

 

Since we are integrating this in the same network (LAN), do we need a SBC? Also I think SBC is inbuilt in PBXNSIP. Right?

 

Should i try using a SBC or is there any other settings we need to do from our side. Below is the logs captured from PBXNSIP.

 

Well, obviously there is not much coming back from 192.168.38.20 on port 5061 (TLS). So I would try outbound proxy sip:192.168.38.20:5060;transport=tcp. You don't need a SBC, that job is already done by the PBX.

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

×
×
  • Create New...