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Konfiguration SIP Direkt?


Tom Tom
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Hallo Zusammen,

 

ich habe ein Problem mit einem SIP Trunk von der Firma Colt Telecom. Diese haben mit nur eine IP mitgegeben und das soll dann wohl direkt auf deeren Border Controller laufen (heißt das so) ich bekomme das allerdings nicht in der Snom konfiguriert.

 

Einstellungen als Gateway mit der IP als SIP Proxy bereits getrestet, bekomme aber immer die Fehlermeldung:

 

Dialplan "Standard Schema": Match 01721009776@localhost to <sip:01721009776@217.110.34.74;user=phone> on trunk Colt

[5] 2012/03/30 11:29:27: INVITE Response 403 Forbidden: Terminate 51988a3d@pbx

 

weiß jemand eine Lösung?

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Hier das Log nach Anleitung

 

[9] 2012/03/30 12:35:08: Last message repeated 2 times

[7] 2012/03/30 12:35:08: SIP Rx tls:10.10.0.50:2629:

REGISTER sip:pbx.company.com SIP/2.0

Via: SIP/2.0/TLS 10.10.0.50:2629;branch=z9hG4bK-mc1995cjtofo;rport

From: "Forty One" <sip:41@pbx.company.com>;tag=8pj4af5nqe

To: "Forty One" <sip:41@pbx.company.com>

Call-ID: 3c26702249e2-bi3qpnp77vpo

CSeq: 233216 REGISTER

Max-Forwards: 70

Contact: <sip:41@10.10.0.50:2629;transport=tls;line=xdtdbgrm>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:a9c7202a-7651-4857-bad0-2b032eb6bfc7>"

User-Agent: snom320/8.4.18

Allow-Events: dialog

X-Real-IP: 10.10.0.50

Supported: path, gruu

WWW-Contact: <http://10.10.0.50:80>

WWW-Contact: <https://10.10.0.50:443>

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2012/03/30 12:35:08: Packet authenticated by transport layer

[7] 2012/03/30 12:35:08: SIP Tx tls:10.10.0.50:2629:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.10.0.50:2629;branch=z9hG4bK-mc1995cjtofo;rport=2629

From: "Forty One" <sip:41@pbx.company.com>;tag=8pj4af5nqe

To: "Forty One" <sip:41@pbx.company.com>;tag=cf8d036600

Call-ID: 3c26702249e2-bi3qpnp77vpo

CSeq: 233216 REGISTER

Contact: <sip:41@10.10.0.50:2629;transport=tls;line=xdtdbgrm>;expires=179

Supported: path

Content-Length: 0

 

 

[7] 2012/03/30 12:35:09: SIP Rx tls:10.10.0.49:2059:

SUBSCRIBE sip:10.10.0.11:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-sbpud3410dc2;rport

From: <sip:49@localhost>;tag=6ny6hja4qh

To: <sip:49@localhost;user=phone>;tag=226a6820c5

Call-ID: 3c267023801b-gdsys4unv55b

CSeq: 34 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:49@10.10.0.49:2059;transport=tls;line=yoxynt0d>;reg-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom320/8.4.18

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2012/03/30 12:35:09: Packet authenticated by transport layer

[7] 2012/03/30 12:35:09: SIP Tx tls:10.10.0.49:2059:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-sbpud3410dc2;rport=2059

From: <sip:49@localhost>;tag=6ny6hja4qh

To: <sip:49@localhost;user=phone>;tag=226a6820c5

Call-ID: 3c267023801b-gdsys4unv55b

CSeq: 34 SUBSCRIBE

Contact: <sip:10.10.0.11:5061;transport=tls>

Expires: 182

Content-Length: 0

 

 

[7] 2012/03/30 12:35:10: SIP Rx tls:10.10.0.49:2059:

INVITE sip:01721009776@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport

From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg

To: <sip:01721009776@localhost;user=phone>

Call-ID: 3c267bc0976c-ccg452e9pmln

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:49@10.10.0.49:2059;transport=tls;line=yoxynt0d>;reg-id=1

X-Serialnumber: 0004133800E1

P-Key-Flags: keys="3"

User-Agent: snom320/8.4.18

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

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 520

 

v=0

o=root 2107876324 2107876324 IN IP4 10.10.0.49

s=call

c=IN IP4 10.10.0.49

t=0 0

m=audio 58144 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZbHe/ofcDrAp8sW5MGIOmEgsfHFJnT1usyc/STE0

a=rtpmap:9 g722/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

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

a=sendrecv

 

[8] 2012/03/30 12:35:10: Packet authenticated by transport layer

[8] 2012/03/30 12:35:10: Allocating call port 62, SIP call id 3c267bc0976c-ccg452e9pmln

[9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:61556

[9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:57720

[9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:62340

[9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:62341

[9] 2012/03/30 12:35:10: UDP: Opening socket on [::]:62340

[9] 2012/03/30 12:35:10: UDP: Opening socket on [::]:62341

[8] 2012/03/30 12:35:10: Could not find a trunk (2 trunks)

[9] 2012/03/30 12:35:10: Using outbound proxy sip:10.10.0.49:2059;transport=tls because of flow-label

[9] 2012/03/30 12:35:10: Last message repeated 3 times

[7] 2012/03/30 12:35:10: SIP Tx tls:10.10.0.49:2059:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport=2059

From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg

To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec

Call-ID: 3c267bc0976c-ccg452e9pmln

CSeq: 1 INVITE

Content-Length: 0

 

 

[7] 2012/03/30 12:35:10: Set packet length to 20

[6] 2012/03/30 12:35:10: Sending RTP for 3c267bc0976c-ccg452e9pmln to 10.10.0.49:58144, codec not set yet

[8] 2012/03/30 12:35:10: Incoming call: Request URI sip:01721009776@localhost;user=phone, To is <sip:01721009776@localhost;user=phone>

[8] 2012/03/30 12:35:10: Call from an user 49

[8] 2012/03/30 12:35:10: To is <sip:01721009776@localhost;user=phone>, user 0, domain 1

[8] 2012/03/30 12:35:10: From user 49

[8] 2012/03/30 12:35:10: Set the To domain based on From user 49@localhost

[8] 2012/03/30 12:35:10: Call state for call object 27: idle

[9] 2012/03/30 12:35:10: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 01721009776@localhost

[5] 2012/03/30 12:35:10: Dialplan "Standard Schema": Match 01721009776@localhost to <sip:01721009776@217.110.34.74;user=phone> on trunk Colt

[8] 2012/03/30 12:35:10: Play audio_moh/noise.wav

[7] 2012/03/30 12:35:10: set_codecs: for 3c267bc0976c-ccg452e9pmln codecs "", codec_preference count 6

[8] 2012/03/30 12:35:10: Allocating call port 63, SIP call id 9d46514b@pbx

[9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:59700

[9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:59701

[9] 2012/03/30 12:35:10: UDP: Opening socket on [::]:59700

[9] 2012/03/30 12:35:10: UDP: Opening socket on [::]:59701

[7] 2012/03/30 12:35:10: set_codecs: for 9d46514b@pbx codecs "", codec_preference count 6

[8] 2012/03/30 12:35:10: call port 63: state code from 0 to 100

[9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: adding codec pcmu/8000 to available list

[9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: adding codec pcma/8000 to available list

[9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: adding codec g722/8000 to available list

[9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: adding codec g726-32/8000 to available list

[9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: adding codec gsm/8000 to available list

[9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: codec_preference size 6, available codecs size 6

[9] 2012/03/30 12:35:10: Resolve 45128: url sip:217.110.34.74

[9] 2012/03/30 12:35:10: Resolve 45128: udp 217.110.34.74 5060

[7] 2012/03/30 12:35:10: SIP Tx udp:217.110.34.74:5060:

INVITE sip:00491721009776@217.110.34.74;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;rport

From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943

To: <sip:00491721009776@217.110.34.74;user=phone>

Call-ID: 9d46514b@pbx

CSeq: 10383 INVITE

Max-Forwards: 70

Contact: <sip:703173588@10.10.0.11:5060;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/2011-4.3.0.5021

Content-Type: application/sdp

Content-Length: 323

 

v=0

o=- 34232 34232 IN IP4 10.10.0.11

s=-

c=IN IP4 10.10.0.11

t=0 0

m=audio 59700 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=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[7] 2012/03/30 12:35:10: SIP Rx tls:10.10.0.27:2591:

SUBSCRIBE sip:10.10.0.11:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 10.10.0.27:2591;branch=z9hG4bK-psxzkrknaqfc;rport

From: <sip:42@pbx.company.com>;tag=4njk5y6n67

To: <sip:42@pbx.company.com;user=phone>;tag=20b41650fd

Call-ID: 3cb37e180a0b-alvbnft4ukno

CSeq: 101800 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:42@10.10.0.27:2591;transport=tls;line=zsuhx64g>;reg-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom320/8.4.18

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2012/03/30 12:35:10: Packet authenticated by transport layer

[7] 2012/03/30 12:35:10: SIP Tx tls:10.10.0.27:2591:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.10.0.27:2591;branch=z9hG4bK-psxzkrknaqfc;rport=2591

From: <sip:42@pbx.company.com>;tag=4njk5y6n67

To: <sip:42@pbx.company.com;user=phone>;tag=20b41650fd

Call-ID: 3cb37e180a0b-alvbnft4ukno

CSeq: 101800 SUBSCRIBE

Contact: <sip:10.10.0.11:5061;transport=tls>

Expires: 179

Content-Length: 0

 

 

[8] 2012/03/30 12:35:10: call port 62: state code from 0 to 183

[7] 2012/03/30 12:35:10: Set packet length to 20

[9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: adding codec pcmu/8000 to available list

[9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: adding codec pcma/8000 to available list

[9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: adding codec g722/8000 to available list

[9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: adding codec g726-32/8000 to available list

[9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: adding codec gsm/8000 to available list

[9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: codec_preference size 6, available codecs size 6

[6] 2012/03/30 12:35:10: Codec pcmu/8000 is chosen for call id 3c267bc0976c-ccg452e9pmln

[7] 2012/03/30 12:35:10: SIP Tx tls:10.10.0.49:2059:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport=2059

From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg

To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec

Call-ID: 3c267bc0976c-ccg452e9pmln

CSeq: 1 INVITE

Contact: <sip:49@10.10.0.11:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.3.0.5021

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 417

 

v=0

o=- 2505 2505 IN IP4 10.10.0.11

s=-

c=IN IP4 10.10.0.11

t=0 0

m=audio 62340 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:GrNA0lN6ROxjsRLIZVwmZ7edO8VjfxIdM0ek+obz

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

a=sendrecv

 

[7] 2012/03/30 12:35:10: SIP Rx udp:217.110.34.74:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;received=213.61.108.147;rport=22980

From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943

To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5

Call-ID: 9d46514b@pbx

CSeq: 10383 INVITE

Content-Length: 0

 

 

[9] 2012/03/30 12:35:10: Message repetition, packet dropped

[7] 2012/03/30 12:35:10: SIP Rx tls:10.10.0.49:2059:

PRACK sip:49@10.10.0.11:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-ijk623j7njd5;rport

From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg

To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec

Call-ID: 3c267bc0976c-ccg452e9pmln

CSeq: 2 PRACK

Max-Forwards: 70

Contact: <sip:49@10.10.0.49:2059;transport=tls;line=yoxynt0d>;reg-id=1

RAck: 1 1 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

 

 

[8] 2012/03/30 12:35:10: Packet authenticated by transport layer

[7] 2012/03/30 12:35:10: SIP Tx tls:10.10.0.49:2059:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-ijk623j7njd5;rport=2059

From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg

To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec

Call-ID: 3c267bc0976c-ccg452e9pmln

CSeq: 2 PRACK

Contact: <sip:49@10.10.0.11:5061;transport=tls>

User-Agent: snom-PBX/2011-4.3.0.5021

Content-Length: 0

 

 

[8] 2012/03/30 12:35:10: SRTP MAC mismatch: f9318abd != 4f4d0000

[7] 2012/03/30 12:35:10: Discard SRTCP packet from 10.10.0.49:58145 with wrong MAC

[7] 2012/03/30 12:35:11: SIP Rx udp:217.110.34.74:5060:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;received=213.61.108.147;rport=22980

From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943

To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5

Call-ID: 9d46514b@pbx

CSeq: 10383 INVITE

Contact: <sip:00491721009776@217.110.34.74:5060>

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

Require: 100rel

RSeq: 14218

Content-Length: 235

Content-Disposition: session; handling=required

Content-Type: application/sdp

 

v=0

o=Sonus_UAC 28998 14816 IN IP4 217.110.34.74

s=SIP Media Capabilities

c=IN IP4 217.110.34.73

t=0 0

m=audio 25878 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

a=ptime:20

 

[7] 2012/03/30 12:35:11: Set packet length to 20

[6] 2012/03/30 12:35:11: Codec pcma/8000 is chosen for call id 9d46514b@pbx

[6] 2012/03/30 12:35:11: Sending RTP for 9d46514b@pbx to 217.110.34.73:25878, codec pcma/8000

[9] 2012/03/30 12:35:11: Resolve 45132: url sip:00491721009776@217.110.34.74:5060

[9] 2012/03/30 12:35:11: Resolve 45132: udp 217.110.34.74 5060

[7] 2012/03/30 12:35:11: SIP Tx udp:217.110.34.74:5060:

PRACK sip:00491721009776@217.110.34.74:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-2c16c59aa7f2b83e02538a5092ae3c3c;rport

From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943

To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5

Call-ID: 9d46514b@pbx

CSeq: 10384 PRACK

Max-Forwards: 70

Contact: <sip:703173588@10.10.0.11:5060;transport=udp>

RAck: 14218 10383 INVITE

Content-Length: 0

 

 

[8] 2012/03/30 12:35:11: Call state for call object 27: alerting

[8] 2012/03/30 12:35:11: call port 62: state code from 183 to 183

[8] 2012/03/30 12:35:11: Last message repeated 2 times

[7] 2012/03/30 12:35:11: 3c267bc0976c-ccg452e9pmln: RTP pass-through mode

[7] 2012/03/30 12:35:11: 9d46514b@pbx: RTP pass-through mode

[6] 2012/03/30 12:35:11: Different Codecs (local pcmu/8000, remote pcma/8000), callid 3c267bc0976c-ccg452e9pmln, falling back to transcoding

[7] 2012/03/30 12:35:11: SIP Rx udp:217.110.34.74:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-2c16c59aa7f2b83e02538a5092ae3c3c;received=213.61.108.147;rport=22980

From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943

To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5

Call-ID: 9d46514b@pbx

CSeq: 10384 PRACK

Content-Length: 0

 

 

[7] 2012/03/30 12:35:11: Call 9d46514b@pbx: Clear last request

[7] 2012/03/30 12:35:12: SIP Rx tls:10.10.0.49:2059:

CANCEL sip:01721009776@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport

From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg

To: <sip:01721009776@localhost;user=phone>

Call-ID: 3c267bc0976c-ccg452e9pmln

CSeq: 1 CANCEL

Max-Forwards: 70

Reason: SIP;cause=487;text="Request terminated by user"

Proxy-Require: buttons

Content-Length: 0

 

 

[8] 2012/03/30 12:35:12: Packet authenticated by transport layer

[7] 2012/03/30 12:35:12: SIP Tx tls:10.10.0.49:2059:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport=2059

From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg

To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec

Call-ID: 3c267bc0976c-ccg452e9pmln

CSeq: 1 CANCEL

Contact: <sip:49@10.10.0.11:5061;transport=tls>

User-Agent: snom-PBX/2011-4.3.0.5021

Content-Length: 0

 

 

[7] 2012/03/30 12:35:12: SIP Tx tls:10.10.0.49:2059:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport=2059

From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg

To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec

Call-ID: 3c267bc0976c-ccg452e9pmln

CSeq: 1 INVITE

Contact: <sip:49@10.10.0.11:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.3.0.5021

Content-Length: 0

 

 

[8] 2012/03/30 12:35:12: Remove leg 50: call port 62, SIP call id 3c267bc0976c-ccg452e9pmln

[8] 2012/03/30 12:35:12: call port 63: state code from 100 to 486

[9] 2012/03/30 12:35:12: Resolve 45135: udp 217.110.34.74 5060

[7] 2012/03/30 12:35:12: SIP Tx udp:217.110.34.74:5060:

CANCEL sip:00491721009776@217.110.34.74;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;rport

From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943

To: <sip:00491721009776@217.110.34.74;user=phone>

Call-ID: 9d46514b@pbx

CSeq: 10383 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[7] 2012/03/30 12:35:12: SIP Rx udp:217.110.34.74:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;received=213.61.108.147;rport=22980

From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943

To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5

Call-ID: 9d46514b@pbx

CSeq: 10383 CANCEL

Content-Length: 0

 

 

[7] 2012/03/30 12:35:12: Call 9d46514b@pbx: Clear last request

[7] 2012/03/30 12:35:12: SIP Rx udp:217.110.34.74:5060:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;received=213.61.108.147;rport=22980

From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943

To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5

Call-ID: 9d46514b@pbx

CSeq: 10383 INVITE

Content-Length: 0

 

 

[7] 2012/03/30 12:35:12: Call 9d46514b@pbx: Clear last INVITE

[9] 2012/03/30 12:35:12: Resolve 45136: url sip:00491721009776@217.110.34.74:5060

[9] 2012/03/30 12:35:12: Resolve 45136: udp 217.110.34.74 5060

[7] 2012/03/30 12:35:12: SIP Tx udp:217.110.34.74:5060:

ACK sip:00491721009776@217.110.34.74:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;rport

From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943

To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5

Call-ID: 9d46514b@pbx

CSeq: 10383 ACK

Max-Forwards: 70

Contact: <sip:703173588@10.10.0.11:5060;transport=udp>

Content-Length: 0

 

 

[5] 2012/03/30 12:35:12: INVITE Response 487 Request Terminated: Terminate 9d46514b@pbx

[7] 2012/03/30 12:35:12: 3c267bc0976c-ccg452e9pmln: Media-aware pass-through mode

[8] 2012/03/30 12:35:12: Clearing call port 63, SIP call id 9d46514b@pbx

[8] 2012/03/30 12:35:12: Remove leg 51: call port 63, SIP call id 9d46514b@pbx

[7] 2012/03/30 12:35:12: SIP Rx tls:10.10.0.49:2059:

ACK sip:01721009776@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport

From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg

To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec

Call-ID: 3c267bc0976c-ccg452e9pmln

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:49@10.10.0.49:2059;transport=tls;line=yoxynt0d>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

 

 

[8] 2012/03/30 12:35:12: Packet authenticated by transport layer

[8] 2012/03/30 12:35:12: Hangup: Call 62 not found

[8] 2012/03/30 12:35:12: Clearing call port 62, SIP call id 3c267bc0976c-ccg452e9pmln

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