Jump to content

Call Transfer didnt with new pbxnsip snom version 4.0


abhisheks
 Share

Recommended Posts

Hi Everyone,

 

We run a production server which makes automatically calls to customers we use Microsoft Speech Server 2007 and PBXNSIP Version 4.0

calls are running properly but when customer care wants to transfer the call to customer care by pressing a key it gets disconnected

firstly i thought it was a code error but it was not because we test with PBXNSIP Version 3.0 it works fine

 

The following log i recieved from PBXNSIP

 

[7] 2011/07/21 20:46:25: Call 63eda295-e385-4cf8-beea-9c21e8272aff: Clear last request

[5] 2011/07/21 20:46:25: BYE Response: Terminate 63eda295-e385-4cf8-beea-9c21e8272aff

[8] 2011/07/21 20:46:25: HTTP client: Connect to 192.168.2.95:80, pending requests 1

[8] 2011/07/21 20:46:25: HTTP client: Connect to 192.168.2.95:80, pending requests 0

[8] 2011/07/21 20:50:36: Release SIP thread 555

[8] 2011/07/21 21:01:36: Release SIP thread 556

[8] 2011/07/22 00:00:10: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:00:10: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:00:10: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:00:10: DNS: Add A exch1.cleard.local 192.168.2.83 (ttl=1200)

[8] 2011/07/22 00:01:10: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:01:11: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:01:11: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:02:11: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:02:12: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:02:12: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:03:12: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:03:12: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:03:12: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:04:12: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:04:13: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:04:13: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:05:13: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:05:14: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:05:14: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:06:14: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:06:14: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:06:14: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:07:14: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:07:15: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:07:15: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:08:15: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:08:16: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:08:16: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:09:16: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:09:17: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:09:17: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:10:17: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:10:17: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:10:17: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:11:17: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:11:18: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:11:18: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:12:18: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:12:19: DNS: Request exch1.cleard.local from server 192.168.2.107

[8] 2011/07/22 00:12:19: DNS: Add AAAA exch1.cleard.local (ttl=60)

[8] 2011/07/22 00:13:19: DNS: AAAA exch1.cleard.local expired

[8] 2011/07/22 00:20:10: DNS: A exch1.cleard.local expired

[8] 2011/07/22 05:34:44: Received SIP connection 557 from 192.168.2.95:4026

[1] 2011/07/22 05:34:44: SIP Rx tcp:192.168.2.95:4026:

INVITE sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone SIP/2.0

FROM: <sip:5145551212@PWD.cleard.local:2431;user=phone>;epid=5FF5C28857;tag=e4c8daf23

TO: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>

CSEQ: 1 INVITE

CALL-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.2.95:4026;branch=z9hG4bKabc5c6a7

CONTACT: <sip:PWD.cleard.local:2431;transport=Tcp;maddr=192.168.2.95;ms-opaque=6815480f729b4e4c>;automata

CONTENT-LENGTH: 335

USER-AGENT: RTCC/3.0.0.0

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 192.168.2.95

s=Microsoft Speech Server session

c=IN IP4 192.168.2.95

t=0 0

m=audio 6272 RTP/AVP 114 115 4 0 8 97 101

a=rtpmap:114 x-msrta/16000

a=fmtp:114 bitrate=29000

a=rtpmap:115 x-msrta/8000

a=fmtp:115 bitrate=11800

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

[1] 2011/07/22 05:34:44: SIP Tx tcp:192.168.2.95:4026:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.2.95:4026;branch=z9hG4bKabc5c6a7

From: <sip:5145551212@PWD.cleard.local:2431;user=phone>;tag=e4c8daf23;epid=5FF5C28857

To: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>;tag=3233189668

Call-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

CSeq: 1 INVITE

Content-Length: 0

 

[7] 2011/07/22 05:34:44: Set packet length to 20

[6] 2011/07/22 05:34:44: Sending RTP for 306f5046-42ca-4688-b7ce-2cbe9156dd1e to 192.168.2.95:6272, codec not set yet

[8] 2011/07/22 05:34:44: Incoming call: Request URI sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone, To is <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>

[8] 2011/07/22 05:34:44: Call from an user 5145551212

[8] 2011/07/22 05:34:44: From user 5145551212

[8] 2011/07/22 05:34:44: Set the To domain based on From user 5145551212@pbx.company.com

[8] 2011/07/22 05:34:44: Call state for call object 5279: idle

[5] 2011/07/22 05:34:44: Dialplan "All Trunks": Match 1_679145_1_5143123784@192.168.2.109 to <sip:15143123784@205.205.211.110> on trunk ISPTELECOM_LD

[8] 2011/07/22 05:34:44: Play audio_moh/noise.wav

[7] 2011/07/22 05:34:44: set_codecs: for 306f5046-42ca-4688-b7ce-2cbe9156dd1e codecs "0 8 9", codec_preference count 4

[7] 2011/07/22 05:34:44: set_codecs: for af91989e@pbx codecs "0 8", codec_preference count 3

[8] 2011/07/22 05:34:44: call port 339: state code from 0 to 100

[1] 2011/07/22 05:34:44: SIP Tx udp:205.205.211.110:5060:

INVITE sip:15143123784@205.205.211.110 SIP/2.0

Via: SIP/2.0/UDP 173.246.64.187:5060;branch=z9hG4bK-0da72cfd2deae66d97f151128b22ef8d;rport

From: "Support" <sip:Anonymous@pbx.company.com;user=phone>;tag=19370

To: <sip:15143123784@205.205.211.110>

Call-ID: af91989e@pbx

CSeq: 1521 INVITE

Max-Forwards: 70

Contact: <sip:Anonymous@173.246.64.187:5060;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/2011-4.2.1.4025

Remote-Party-ID: "PWD cleard.local" <sip:Anonymous@pbx.company.com;user=phone>;party=calling;screen=yes

Content-Type: application/sdp

Content-Length: 257

 

v=0

o=- 34200 34200 IN IP4 173.246.64.187

s=-

c=IN IP4 173.246.64.187

t=0 0

m=audio 51694 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

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

a=sendrecv

[8] 2011/07/22 05:34:44: call port 338: state code from 0 to 183

[7] 2011/07/22 05:34:44: Set packet length to 20

[6] 2011/07/22 05:34:44: Codec pcmu/8000 is chosen for call id 306f5046-42ca-4688-b7ce-2cbe9156dd1e

[1] 2011/07/22 05:34:44: SIP Tx tcp:192.168.2.95:4026:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.2.95:4026;branch=z9hG4bKabc5c6a7

From: <sip:5145551212@PWD.cleard.local:2431;user=phone>;tag=e4c8daf23;epid=5FF5C28857

To: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>;tag=3233189668

Call-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

CSeq: 1 INVITE

Contact: <sip:5145551212@192.168.2.109:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2011-4.2.1.4025

Content-Type: application/sdp

Content-Length: 267

 

v=0

o=- 15799 15799 IN IP4 192.168.2.109

s=-

c=IN IP4 192.168.2.109

t=0 0

m=audio 60466 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/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

[1] 2011/07/22 05:34:44: SIP Rx udp:205.205.211.110:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 173.246.64.187:5060;branch=z9hG4bK-0da72cfd2deae66d97f151128b22ef8d;rport

From: "Support" <sip:Anonymous@pbx.company.com;user=phone>;tag=19370

To: <sip:15143123784@205.205.211.110>

Call-ID: af91989e@pbx

CSeq: 1521 INVITE

Content-Length: 0

 

[6] 2011/07/22 05:34:47: Sending RTP for af91989e@pbx to 209.167.192.3:10006, codec not set yet

[1] 2011/07/22 05:34:47: SIP Rx udp:205.205.211.110:5060:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 173.246.64.187:5060;branch=z9hG4bK-0da72cfd2deae66d97f151128b22ef8d;rport

From: "Support" <sip:Anonymous@pbx.company.com;user=phone>;tag=19370

To: <sip:15143123784@205.205.211.110>;tag=zIl9009r3a9zyad-IPTrunk-898-19-21at205.205.211.110

Call-ID: af91989e@pbx

CSeq: 1521 INVITE

Allow: REGISTER,INVITE,CANCEL,BYE,ACK

Content-Type: application/sdp

Content-Length: 195

 

v=0

o=- 1 1 IN IP4 205.205.211.110

s=SIP Call

c=IN IP4 209.167.192.3

t=0 0

m=audio 10006 RTP/AVP 0 101

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:0 PCMU/8000

[7] 2011/07/22 05:34:47: Set packet length to 20

[6] 2011/07/22 05:34:47: Codec pcmu/8000 is chosen for call id af91989e@pbx

[8] 2011/07/22 05:34:47: Call state for call object 5279: alerting

[8] 2011/07/22 05:34:47: call port 338: state code from 183 to 183

[8] 2011/07/22 05:34:47: Last message repeated 2 times

[7] 2011/07/22 05:34:47: 306f5046-42ca-4688-b7ce-2cbe9156dd1e: RTP pass-through mode

[7] 2011/07/22 05:34:47: af91989e@pbx: RTP pass-through mode

[1] 2011/07/22 05:34:50: SIP Rx udp:205.205.211.110:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 173.246.64.187:5060;branch=z9hG4bK-0da72cfd2deae66d97f151128b22ef8d;rport

From: "Support" <sip:Anonymous@pbx.company.com;user=phone>;tag=19370

To: <sip:15143123784@205.205.211.110>;tag=zIl9009r3a9zyad-IPTrunk-898-19-21at205.205.211.110

Call-ID: af91989e@pbx

CSeq: 1521 INVITE

Contact: sip:205.205.211.110:5060

Content-Type: application/sdp

Content-Length: 195

 

v=0

o=- 1 1 IN IP4 205.205.211.110

s=SIP Call

c=IN IP4 209.167.192.3

t=0 0

m=audio 10006 RTP/AVP 0 101

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:0 PCMU/8000

[7] 2011/07/22 05:34:50: Call af91989e@pbx: Clear last INVITE

[7] 2011/07/22 05:34:50: Set packet length to 20

[1] 2011/07/22 05:34:50: SIP Tx udp:205.205.211.110:5060:

ACK sip:205.205.211.110:5060 SIP/2.0

Via: SIP/2.0/UDP 173.246.64.187:5060;branch=z9hG4bK-6acd7ccc3a1bf84a9783f6c8bd8ea5a0;rport

From: "Support" <sip:Anonymous@pbx.company.com;user=phone>;tag=19370

To: <sip:15143123784@205.205.211.110>;tag=zIl9009r3a9zyad-IPTrunk-898-19-21at205.205.211.110

Call-ID: af91989e@pbx

CSeq: 1521 ACK

Max-Forwards: 70

Contact: <sip:Anonymous@173.246.64.187:5060;transport=udp>

Remote-Party-ID: "Support" <sip:Anonymous@pbx.company.com;user=phone>;party=calling;screen=yes

Content-Length: 0

 

[7] 2011/07/22 05:34:50: Determine pass-through mode after receiving response

[8] 2011/07/22 05:34:50: Call state for call object 5279: connected

[8] 2011/07/22 05:34:50: call port 339: state code from 100 to 200

[8] 2011/07/22 05:34:50: call port 338: state code from 183 to 200

[1] 2011/07/22 05:34:50: SIP Tx tcp:192.168.2.95:4026:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.2.95:4026;branch=z9hG4bKabc5c6a7

From: <sip:5145551212@PWD.cleard.local:2431;user=phone>;tag=e4c8daf23;epid=5FF5C28857

To: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>;tag=3233189668

Call-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

CSeq: 1 INVITE

Contact: <sip:5145551212@192.168.2.109:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2011-4.2.1.4025

Content-Type: application/sdp

Content-Length: 267

 

v=0

o=- 15799 15799 IN IP4 192.168.2.109

s=-

c=IN IP4 192.168.2.109

t=0 0

m=audio 60466 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/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

[1] 2011/07/22 05:34:51: SIP Rx tcp:192.168.2.95:4026:

ACK sip:5145551212@192.168.2.109:5060;transport=tcp SIP/2.0

FROM: <sip:5145551212@PWD.cleard.local:2431;user=phone>;epid=5FF5C28857;tag=e4c8daf23

TO: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>;tag=3233189668

CSEQ: 1 ACK

CALL-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.2.95:4026;branch=z9hG4bK32dd5b

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.0.0.0

 

[7] 2011/07/22 05:35:08: Received RFC4733 DTMF on codec 101

[1] 2011/07/22 05:35:48: SIP Rx tcp:192.168.2.95:4026:

REFER sip:5145551212@192.168.2.109:5060;transport=tcp SIP/2.0

FROM: <sip:5145551212@PWD.cleard.local:2431;user=phone>;epid=5FF5C28857;tag=e4c8daf23

TO: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>;tag=3233189668

CSEQ: 2 REFER

CALL-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.2.95:4026;branch=z9hG4bK3e1f596

CONTACT: <sip:PWD.cleard.local:2431;transport=Tcp;maddr=192.168.2.95;ms-opaque=6815480f729b4e4c>;automata

CONTENT-LENGTH: 0

REFER-TO: <sip:1_679145_1_8885250930@192.168.2.109:5060;transport=tcp;user=phone>

REFERRED-BY: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>

USER-AGENT: RTCC/3.0.0.0

 

[1] 2011/07/22 05:35:48: SIP Tx tcp:192.168.2.95:4026:

SIP/2.0 202 Accepted

Via: SIP/2.0/TCP 192.168.2.95:4026;branch=z9hG4bK3e1f596

From: <sip:5145551212@PWD.cleard.local:2431;user=phone>;tag=e4c8daf23;epid=5FF5C28857

To: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>;tag=3233189668

Call-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

CSeq: 2 REFER

Contact: <sip:5145551212@192.168.2.109:5060;transport=tcp>

User-Agent: pbxnsip-PBX/2011-4.2.1.4025

Content-Length: 0

 

[1] 2011/07/22 05:35:48: SIP Tx tcp:192.168.2.95:4026:

NOTIFY sip:PWD.cleard.local:2431;transport=Tcp;maddr=192.168.2.95;ms-opaque=6815480f729b4e4c SIP/2.0

Via: SIP/2.0/TCP 192.168.2.109:5060;branch=z9hG4bK-ede670ecb13735b24cfefcdf3806f002;rport

From: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>;tag=3233189668

To: <sip:5145551212@PWD.cleard.local:2431;user=phone>;tag=e4c8daf23;epid=5FF5C28857

Call-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

CSeq: 30309 NOTIFY

Max-Forwards: 70

Contact: <sip:5145551212@192.168.2.109:5060;transport=tcp>

Subscription-State: terminated;reason=noresource

Event: refer

Content-Type: message/sipfrag

Content-Length: 16

 

SIP/2.0 200 Ok

[7] 2011/07/22 05:35:48: REFER from device type "RTCC/3.0.0.0" is not supported in this product

[8] 2011/07/22 05:35:48: call port 338: state code from 200 to 486

[5] 2011/07/22 05:35:48: Call 306f5046-42ca-4688-b7ce-2cbe9156dd1e: Last request not finished

[1] 2011/07/22 05:35:48: SIP Tx tcp:192.168.2.95:4026:

BYE sip:PWD.cleard.local:2431;transport=Tcp;maddr=192.168.2.95;ms-opaque=6815480f729b4e4c SIP/2.0

Via: SIP/2.0/TCP 192.168.2.109:5060;branch=z9hG4bK-1d8828e895b50439cb974282777e7394;rport

From: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>;tag=3233189668

To: <sip:5145551212@PWD.cleard.local:2431;user=phone>;tag=e4c8daf23;epid=5FF5C28857

Call-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

CSeq: 30310 BYE

Max-Forwards: 70

Contact: <sip:5145551212@192.168.2.109:5060;transport=tcp>

Content-Length: 0

 

[8] 2011/07/22 05:35:48: call port 339: state code from 200 to 486

[1] 2011/07/22 05:35:48: SIP Tx udp:205.205.211.110:5060:

BYE sip:205.205.211.110:5060 SIP/2.0

Via: SIP/2.0/UDP 173.246.64.187:5060;branch=z9hG4bK-3e2ee8c5a1e32389e183ce53b16466a6;rport

From: "Support" <sip:Anonymous@pbx.company.com;user=phone>;tag=19370

To: <sip:15143123784@205.205.211.110>;tag=zIl9009r3a9zyad-IPTrunk-898-19-21at205.205.211.110

Call-ID: af91989e@pbx

CSeq: 1522 BYE

Max-Forwards: 70

Contact: <sip:Anonymous@173.246.64.187:5060;transport=udp>

Remote-Party-ID: "PWD cleard.local" <sip:Anonymous@pbx.company.com;user=phone>;party=calling;screen=yes

Content-Length: 0

 

[1] 2011/07/22 05:35:48: SIP Rx udp:205.205.211.110:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 173.246.64.187:5060;branch=z9hG4bK-3e2ee8c5a1e32389e183ce53b16466a6;rport

From: "Support" <sip:Anonymous@pbx.company.com;user=phone>;tag=19370

To: <sip:15143123784@205.205.211.110>;tag=zIl9009r3a9zyad-IPTrunk-898-19-21at205.205.211.110

Call-ID: af91989e@pbx

CSeq: 1522 BYE

Content-Length: 0

 

[7] 2011/07/22 05:35:48: Call af91989e@pbx: Clear last request

[5] 2011/07/22 05:35:48: BYE Response: Terminate af91989e@pbx

[7] 2011/07/22 05:35:48: 306f5046-42ca-4688-b7ce-2cbe9156dd1e: Media-aware pass-through mode

[8] 2011/07/22 05:35:48: HTTP client: Connect to 192.168.2.95:80, pending requests 0

[1] 2011/07/22 05:35:48: SIP Rx tcp:192.168.2.95:4026:

SIP/2.0 200 OK

FROM: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>;tag=3233189668

TO: <sip:5145551212@PWD.cleard.local:2431;user=phone>;tag=e4c8daf23;epid=5FF5C28857

CSEQ: 30309 NOTIFY

CALL-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

VIA: SIP/2.0/TCP 192.168.2.109:5060;branch=z9hG4bK-ede670ecb13735b24cfefcdf3806f002;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

[8] 2011/07/22 05:35:48: Call 306f5046-42ca-4688-b7ce-2cbe9156dd1e: Response does not correspond to open request

[1] 2011/07/22 05:35:49: SIP Rx tcp:192.168.2.95:4026:

SIP/2.0 200 OK

FROM: <sip:1_679145_1_5143123784@192.168.2.109:5060;user=phone>;tag=3233189668

TO: <sip:5145551212@PWD.cleard.local:2431;user=phone>;tag=e4c8daf23;epid=5FF5C28857

CSEQ: 30310 BYE

CALL-ID: 306f5046-42ca-4688-b7ce-2cbe9156dd1e

VIA: SIP/2.0/TCP 192.168.2.109:5060;branch=z9hG4bK-1d8828e895b50439cb974282777e7394;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

[8] 2011/07/22 05:35:49: Call 306f5046-42ca-4688-b7ce-2cbe9156dd1e: Response does not correspond to open request

[5] 2011/07/22 05:35:49: BYE Response: Terminate 306f5046-42ca-4688-b7ce-2cbe9156dd1e

[8] 2011/07/22 05:35:49: HTTP client: Connect to 192.168.2.95:80, pending requests 0

 

 

Please help its urgent

 

 

 

Thanks,

Abhishek

Link to comment
Share on other sites

[7] 2011/07/22 05:35:48: REFER from device type "RTCC/3.0.0.0" is not supported in this product

 

snom decided to support REFER only from snom products. You need snom ONE green in order to have this feature available. If you have a old pbxnsip license, you can as well use a pbxnsip build which does not have this restriction.

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...