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Snom One Plus & ISDN -> gibt es eine Step-By-Step-Anleitung?


tomcat_34
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Wir haben hier eine Snom One Plus mit 12 ISDN-Ports und diversen Telefonen und bekommen das Ding nicht ans Laufen.

Das englischsprachige Handbuch erklärt m.E. diverse Begriffe nicht ausreichend um die Anlage systematisch konfigurieren zu können.

 

Interne Telefonate sind kein Problem, es geht mir ausschließlich darum, die Verbindung zu den internen ISDN-Karten für ein- und ausgehende Gespräche ans Laufen zu bringen.

Offen gestanden bin ich recht enttäuscht dass hier der Eindruck eines Produktes "aus einer Hand" vermittelt wird und man dann eine doch eher zusammengestückelte Lösung bekommt gerade was die ISDN-Anbindung angeht.

Wenn ich von der Konfigurations-Konsole aus Hilfen aufrufe erhalte ich an allen interessanten Stellen den Hinweis "Hier ist noch kein Text hinterlegt".

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Hallo,

 

Haben Sie das Quick Start schon durchgeführt? Was passiert wenn Sie versuchen zu telefonieren?

 

Benutzen Sie Anlageanschluss oder Mehrgeräteanschluss ? Bitte kontrollieren Sie bei Sangoma -> NetBorder Express Gateway -> Configuration -> PSTN config -> Call control -> ISDN Configurations -> BRI L2 Configuration -> Topology, das sollte POINT-TO-POINT für Anlageanschluss oder MULTIPOINT für Mehrgeräteanschluss stehen.

 

Wenn das nicht hilft können Sie bitte debug aktivieren wie hier erklärt http://wiki.snomone.com/index.php?title=Snom_ONE_log ? Und dann die Logs posten, für ein- und ausgehende Gespräche ?

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Hallo Zusammen,

ich bin im gleichen Dilemma! B700 BRI mit Netborder Express, Installation ohne Fehler abgeschlossen!

Das Netborder Gateway läuft laut Webbrowser. Die vorher genannten Einstellungen an der BRI L2 Configuration habe ich übernommen. Das Logging kann ich über den angegebenen Link nicht aktivieren. Die Version "2011-4.5.0.1030 Beta Corona Austrinids (CentOS32)" hat an der angegebenen Stelle keinen Schalter fürs Logging. Auch ich bin nach 2 Wochen Herumprobieren kurz vor dem "Absprung"!

Ich bitte um Hilfe!

Danke schon mal im Voraus!

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Hier kommt ein Auszug aus meinem Logging:

 

[6] 2012/03/19 16:31:17: Received bindRequest for user pbx.company.com\40

[5] 2012/03/19 16:31:22: SIP Rx tls:192.168.100.28:4894:

INVITE sip:0084581200@pbx.company.com;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:4894;branch=z9hG4bK-1qckjrlzu8vt;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=da5ztnucwh

To: <sip:0084581200@pbx.company.com;user=phone>

Call-ID: dd70263c1dd3-bv404jnuvspz

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:4894;transport=tls;line=0ii3wcy6>;reg-id=1

X-Serialnumber: 000413412514

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

User-Agent: snom870/8.4.18

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: 528

 

v=0

o=root 1437626661 1437626661 IN IP4 192.168.100.28

s=call

c=IN IP4 192.168.100.28

t=0 0

m=audio 60024 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:hxeiS6pr0133Ddk/slWWoXjLxx/zCy9LpoaVoo4W

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] 2012/03/19 16:31:22: Packet authenticated by transport layer

[8] 2012/03/19 16:31:22: Allocating for call port 0, SIP call id dd70263c1dd3-bv404jnuvspz

[9] 2012/03/19 16:31:22: UDP(IPv4): Opening socket on 0.0.0.0:59472

[9] 2012/03/19 16:31:22: UDP(IPv4): Opening socket on 0.0.0.0:59473

[9] 2012/03/19 16:31:22: UDP(IPv6): Opening socket on [::]:59472

[9] 2012/03/19 16:31:22: UDP(IPv6): Opening socket on [::]:59473

[8] 2012/03/19 16:31:22: Could not find a trunk (3 trunks)

[9] 2012/03/19 16:31:22: Using outbound proxy sip:192.168.100.28:4894;transport=tls because of flow-label

[9] 2012/03/19 16:31:22: Last message repeated 3 times

[5] 2012/03/19 16:31:22: SIP Tx tls:192.168.100.28:4894:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.100.28:4894;branch=z9hG4bK-1qckjrlzu8vt;rport=4894

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=da5ztnucwh

To: <sip:0084581200@pbx.company.com;user=phone>;tag=78d741b8bb

Call-ID: dd70263c1dd3-bv404jnuvspz

CSeq: 1 INVITE

Content-Length: 0

 

[6] 2012/03/19 16:31:22: Received searchRequest(type 128), substrings=+84581200

[7] 2012/03/19 16:31:22: Set packet length to 20

[6] 2012/03/19 16:31:22: Call-leg 0: Sending RTP for dd70263c1dd3-bv404jnuvspz to 192.168.100.28:60024, codec not set yet

[8] 2012/03/19 16:31:22: Incoming call: Request URI sip:0084581200@pbx.company.com;user=phone, To is <sip:0084581200@pbx.company.com;user=phone>

[8] 2012/03/19 16:31:22: Call from an user 40

[8] 2012/03/19 16:31:22: To is <sip:0084581200@pbx.company.com;user=phone>, user 0, domain 1

[8] 2012/03/19 16:31:22: From user 40

[8] 2012/03/19 16:31:22: Set the To domain based on From user 40@pbx.company.com

[8] 2012/03/19 16:31:22: Call state for call object 1: idle

[7] 2012/03/19 16:31:22: Call port 0: set_codecs for dd70263c1dd3-bv404jnuvspz codecs "", codec_preference count 6

[9] 2012/03/19 16:31:22: Dialplan: Evaluating !^9([0-9]*)@.*!sip:\1@\r;user=phone!i against 0084581200@pbx.company.com

[6] 2012/03/19 16:31:22: The trunk sipgate is disabled. Skipping it...

[9] 2012/03/19 16:31:22: Dialplan: Evaluating !^0([0-9]*)@.*!sip:\1@\r;user=phone!i against 0084581200@pbx.company.com

[5] 2012/03/19 16:31:22: Dialplan "Hackner": Match 0084581200@pbx.company.com to sip:084581200@127.0.0.1:5066;user=phone on trunk NBE

[9] 2012/03/19 16:31:22: Generating hf header using {from}

[9] 2012/03/19 16:31:22: Generating ht header using {to}

[9] 2012/03/19 16:31:22: Generating hppi header using {trunk}

[8] 2012/03/19 16:31:22: Play audio_moh/noise.wav, caching true

[8] 2012/03/19 16:31:22: Allocating for call port 1, SIP call id da12de4e@pbx

[9] 2012/03/19 16:31:22: UDP(IPv4): Opening socket on 0.0.0.0:61438

[9] 2012/03/19 16:31:22: UDP(IPv4): Opening socket on 0.0.0.0:61439

[9] 2012/03/19 16:31:22: UDP(IPv6): Opening socket on [::]:61438

[9] 2012/03/19 16:31:22: UDP(IPv6): Opening socket on [::]:61439

[7] 2012/03/19 16:31:22: Call port 1: set_codecs for da12de4e@pbx codecs "", codec_preference count 6

[9] 2012/03/19 16:31:22: Call port 1: update_codecs for da12de4e@pbx: adding codec pcmu/8000 to available list

[9] 2012/03/19 16:31:22: Call port 1: update_codecs for da12de4e@pbx: adding codec pcma/8000 to available list

[9] 2012/03/19 16:31:22: Call port 1: update_codecs for da12de4e@pbx: adding codec g722/8000 to available list

[9] 2012/03/19 16:31:22: Call port 1: update_codecs for da12de4e@pbx: adding codec g726-32/8000 to available list

[9] 2012/03/19 16:31:22: Call port 1: update_codecs for da12de4e@pbx: adding codec gsm/8000 to available list

[9] 2012/03/19 16:31:22: Call port 1: update_codecs for da12de4e@pbx: codec_preference size 6, available codecs size 6

[9] 2012/03/19 16:31:22: Resolve 17: url sip:127.0.0.1:5066

[9] 2012/03/19 16:31:22: Resolve 17: udp 127.0.0.1 5066

[5] 2012/03/19 16:31:22: SIP Tx udp:127.0.0.1:5066:

INVITE sip:084581200@127.0.0.1:5066;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-b7db22d8a3878234e37a797478260d02;rport

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=1618927444

To: <sip:0084581200@pbx.company.com;user=phone>

Call-ID: da12de4e@pbx

CSeq: 16731 INVITE

Max-Forwards: 70

Contact: <sip:40@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

P-Preferred-Identity: "pbx.company.com" <sip:>

P-Charging-Vector: icid-value=;icid-generated-at=127.0.0.1;orig-ioi=pbx.company.com

Content-Type: application/sdp

Content-Length: 329

 

v=0

o=- 565391363 565391363 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 61438 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] 2012/03/19 16:31:22: Set packet length to 20

[9] 2012/03/19 16:31:22: Call port 0: update_codecs for dd70263c1dd3-bv404jnuvspz: adding codec pcmu/8000 to available list

[9] 2012/03/19 16:31:22: Call port 0: update_codecs for dd70263c1dd3-bv404jnuvspz: adding codec pcma/8000 to available list

[9] 2012/03/19 16:31:22: Call port 0: update_codecs for dd70263c1dd3-bv404jnuvspz: adding codec g722/8000 to available list

[9] 2012/03/19 16:31:22: Call port 0: update_codecs for dd70263c1dd3-bv404jnuvspz: adding codec g726-32/8000 to available list

[9] 2012/03/19 16:31:22: Call port 0: update_codecs for dd70263c1dd3-bv404jnuvspz: adding codec gsm/8000 to available list

[9] 2012/03/19 16:31:22: Call port 0: update_codecs for dd70263c1dd3-bv404jnuvspz: codec_preference size 6, available codecs size 6

[6] 2012/03/19 16:31:22: Call-leg 0: Codec pcmu/8000 is chosen for call id dd70263c1dd3-bv404jnuvspz

[5] 2012/03/19 16:31:22: set codec: codec pcmu/8000 is set to call-leg 0

[5] 2012/03/19 16:31:22: SIP Tx tls:192.168.100.28:4894:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TLS 192.168.100.28:4894;branch=z9hG4bK-1qckjrlzu8vt;rport=4894

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=da5ztnucwh

To: <sip:0084581200@pbx.company.com;user=phone>;tag=78d741b8bb

Call-ID: dd70263c1dd3-bv404jnuvspz

CSeq: 1 INVITE

Contact: <sip:40@192.168.100.9:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 435

 

v=0

o=- 1225040769 1225040769 IN IP4 192.168.100.9

s=-

c=IN IP4 192.168.100.9

t=0 0

m=audio 59472 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ImOdxFUfFVO9MfAVx8zOpFTVA6B1LuqB8o0/Q2fR

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

[5] 2012/03/19 16:31:22: SIP Rx udp:127.0.0.1:5066:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-b7db22d8a3878234e37a797478260d02;rport=5060

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=1618927444

To: <sip:0084581200@pbx.company.com;user=phone>;tag=ds-1c69fb81-9608a8

Call-ID: da12de4e@pbx

CSeq: 16731 INVITE

Content-Length: 0

Server: Netborder Express Gateway/4.1.6

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

 

[9] 2012/03/19 16:31:22: Message repetition, packet dropped

[6] 2012/03/19 16:31:22: Received searchRequest(type 128), substrings=0084581200

[5] 2012/03/19 16:31:22: SIP Rx tls:192.168.100.28:4894:

PRACK sip:40@192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:4894;branch=z9hG4bK-k6gephoj7htk;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=da5ztnucwh

To: <sip:0084581200@pbx.company.com;user=phone>;tag=78d741b8bb

Call-ID: dd70263c1dd3-bv404jnuvspz

CSeq: 2 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:4894;transport=tls;line=0ii3wcy6>;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] 2012/03/19 16:31:22: Packet authenticated by transport layer

[5] 2012/03/19 16:31:22: SIP Tx tls:192.168.100.28:4894:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:4894;branch=z9hG4bK-k6gephoj7htk;rport=4894

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=da5ztnucwh

To: <sip:0084581200@pbx.company.com;user=phone>;tag=78d741b8bb

Call-ID: dd70263c1dd3-bv404jnuvspz

CSeq: 2 PRACK

Contact: <sip:40@192.168.100.9:5061;transport=tls>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[5] 2012/03/19 16:31:29: SIP Rx tls:192.168.100.28:4894:

CANCEL sip:0084581200@pbx.company.com;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:4894;branch=z9hG4bK-1qckjrlzu8vt;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=da5ztnucwh

To: <sip:0084581200@pbx.company.com;user=phone>

Call-ID: dd70263c1dd3-bv404jnuvspz

CSeq: 1 CANCEL

Max-Forwards: 70

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

Proxy-Require: buttons

Content-Length: 0

 

[8] 2012/03/19 16:31:29: Packet authenticated by transport layer

[5] 2012/03/19 16:31:29: SIP Tx tls:192.168.100.28:4894:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/TLS 192.168.100.28:4894;branch=z9hG4bK-1qckjrlzu8vt;rport=4894

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=da5ztnucwh

To: <sip:0084581200@pbx.company.com;user=phone>;tag=78d741b8bb

Call-ID: dd70263c1dd3-bv404jnuvspz

CSeq: 1 INVITE

Contact: <sip:40@192.168.100.9:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[9] 2012/03/19 16:31:29: Resolve 21: udp 127.0.0.1 5066

[5] 2012/03/19 16:31:29: SIP Tx udp:127.0.0.1:5066:

CANCEL sip:084581200@127.0.0.1:5066;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-b7db22d8a3878234e37a797478260d02;rport

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=1618927444

To: <sip:0084581200@pbx.company.com;user=phone>

Call-ID: da12de4e@pbx

CSeq: 16731 CANCEL

Max-Forwards: 70

P-Preferred-Identity: "pbx.company.com" <sip:>

P-Charging-Vector: icid-value=;icid-generated-at=127.0.0.1;orig-ioi=pbx.company.com

Content-Length: 0

 

[5] 2012/03/19 16:31:29: SIP Rx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-b7db22d8a3878234e37a797478260d02;rport=5060

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=1618927444

To: <sip:0084581200@pbx.company.com;user=phone>

Call-ID: da12de4e@pbx

CSeq: 16731 CANCEL

Content-Length: 0

 

[7] 2012/03/19 16:31:29: Call da12de4e@pbx: Clear last request

[5] 2012/03/19 16:31:29: SIP Rx udp:127.0.0.1:5066:

SIP/2.0 487 Transaction Cancelled

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-b7db22d8a3878234e37a797478260d02;rport=5060

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=1618927444

To: <sip:0084581200@pbx.company.com;user=phone>;tag=ds-1c69fb81-9608a8

Call-ID: da12de4e@pbx

CSeq: 16731 INVITE

Content-Length: 0

Contact: <sip:127.0.0.1:5066;transport=udp>

 

[7] 2012/03/19 16:31:29: Call da12de4e@pbx: Clear last INVITE

[9] 2012/03/19 16:31:29: Resolve 22: url sip:127.0.0.1:5066

[9] 2012/03/19 16:31:29: Resolve 22: udp 127.0.0.1 5066

[5] 2012/03/19 16:31:29: SIP Tx udp:127.0.0.1:5066:

ACK sip:084581200@127.0.0.1:5066;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-b7db22d8a3878234e37a797478260d02;rport

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=1618927444

To: <sip:0084581200@pbx.company.com;user=phone>;tag=ds-1c69fb81-9608a8

Call-ID: da12de4e@pbx

CSeq: 16731 ACK

Max-Forwards: 70

Contact: <sip:40@127.0.0.1:5060;transport=udp>

P-Preferred-Identity: "pbx.company.com" <sip:>

P-Charging-Vector: icid-value=;icid-generated-at=127.0.0.1;orig-ioi=pbx.company.com

Content-Length: 0

 

[5] 2012/03/19 16:31:29: INVITE Response 487 Transaction Cancelled: Terminate da12de4e@pbx

[8] 2012/03/19 16:31:29: Clearing call port 1, SIP call id da12de4e@pbx

[8] 2012/03/19 16:31:29: Remove leg 1: call port 0, SIP call id dd70263c1dd3-bv404jnuvspz

[8] 2012/03/19 16:31:29: Remove leg 2: call port 1, SIP call id da12de4e@pbx

[9] 2012/03/19 16:31:29: Remote site 192.168.100.28 closed the connection

[5] 2012/03/19 16:31:29: SIP Rx tls:192.168.100.28:4894:

ACK sip:0084581200@pbx.company.com;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:4894;branch=z9hG4bK-1qckjrlzu8vt;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=da5ztnucwh

To: <sip:0084581200@pbx.company.com;user=phone>;tag=78d741b8bb

Call-ID: dd70263c1dd3-bv404jnuvspz

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:4894;transport=tls;line=0ii3wcy6>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

 

[8] 2012/03/19 16:31:29: Packet authenticated by transport layer

[8] 2012/03/19 16:31:29: Clearing call port 0, SIP call id dd70263c1dd3-bv404jnuvspz

[8] 2012/03/19 16:31:29: Hangup: Call 0 not found

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Danke für Ihre schnelle Antwort!

Mit "Quick Start" meinen Sie "Quick Setup" im Netborder Gateway Manager? Das habe ich gemacht, wie in "http://wiki.snomone.com/index.php?title=Configuring_NetBorder_Express_With_snom_ONE&direction=next&oldid=4923" geschildert!

Die Einstellung für Anlageanschluss oder Mehrgeräteanschluss werden im Netborder Gateway Manager gemacht? Oder im SnomONE? Die Einstellung im Netborder war für unseren Mehrgeräteanschluß war auf Multipoint!

Muss ich bei :

1) PSTN Config -> Physical Configurations -> P1-digital_BRI_TE -> Digital BRI Configuration -> clocking: Terminal oder Network setzen?

2) PSTN Config -> ISDN Configurations -> BRI1_TE -> Termination: Terminal oder Network setzen?

3) PSTN Config -> Call Control -> Channel Groups: eine neue Gruppe erzeugen, wie in manchen Anleitungen des Internets geraten wird?

4) Muss ich im SnomONE Manager -> Leitungen -> NBE -> Ziel Konto = 70 eintragen?

5) Kann ich irgendwo MSN's hinterlegen?

6) Muss im Netborder Gateway Manager unter Configuration -> Routing Rules irgendetwas getan werden? Wenn ja, bitte ich um Vorlagen dazu.

7) Ist die Passage: [9] 2012/03/19 16:31:22: UDP(IPv4): Opening socket on 0.0.0.0:61438

[9] 2012/03/19 16:31:22: UDP(IPv4): Opening socket on 0.0.0.0:61439

so OK? das Opening socket on 0.0.0.0:61438 beunruhigt mich.

Viele Fragen - hoffentlich viele Antworten!

Danke im Voraus!

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Hallo,

 

>Mit "Quick Start" meinen Sie "Quick Setup" im Netborder Gateway Manager? Das habe ich gemacht, wie in "http://wiki.snomone.com/index.php?title=Configuring_NetBorder_Express_With_snom_ONE&direction=next&oldid=4923" geschildert!

Ja, das meinte ich.

 

>Die Einstellung für Anlageanschluss oder Mehrgeräteanschluss werden im Netborder Gateway Manager gemacht? Oder im SnomONE? Die Einstellung im Netborder war für unseren Mehrgeräteanschluß war auf Multipoint!

Das ist auch richtig, im Netborder muss man Multipoint setzten

 

>Muss ich bei :

>1) PSTN Config -> Physical Configurations -> P1-digital_BRI_TE -> Digital BRI Configuration -> clocking: Terminal oder Network setzen?

Das Quickt Setup sollte das automatisch auf Terminal setzten

 

>2) PSTN Config -> ISDN Configurations -> BRI1_TE -> Termination: Terminal oder Network setzen?

Das Quickt Setup sollte das automatisch auf Terminal setzten

 

>3) PSTN Config -> Call Control -> Channel Groups: eine neue Gruppe erzeugen, wie in manchen Anleitungen des Internets geraten wird?

Da brauchen Sie auch erstmal nichts machen, das Quickt Setup macht es automatisch

 

>4) Muss ich im SnomONE Manager -> Leitungen -> NBE -> Ziel Konto = 70 eintragen?

Ich würde schon erst mal 70 eintragen, um einfacher zu testen, ob eingehende Gespräche funktionieren. Wenn das geht dann können Sie als Ziel Konto auch ein komplexer String benutzen. zB wenn Ihre MSNs 3 Zahlen haben, können Sie die snom ONE Nebestelle gleich nummerieren, und dann als Ziel Konto:

!([0-9]{3}$)!\1!

(mehr Info hier: http://wiki.snomone.com/index.php?title=Inbounds_Calls#How_the_System_Routes_a_Call_to_the_Proper_Extension)

 

>5) Kann ich irgendwo MSN's hinterlegen?

Ja, für jede Nebestelle kann man als ANI die ganze Nummer (Kopfnummer+MSN) eingeben, damit das CallerID bei ausgehende Anrufe stimmt

 

>6) Muss im Netborder Gateway Manager unter Configuration -> Routing Rules irgendetwas getan werden? Wenn ja, bitte ich um Vorlagen dazu.

Das können Sie auch erst mal lassen.

 

>7) Ist die Passage: [9] 2012/03/19 16:31:22: UDP(IPv4): Opening socket on 0.0.0.0:61438

>[9] 2012/03/19 16:31:22: UDP(IPv4): Opening socket on 0.0.0.0:61439

>so OK? das Opening socket on 0.0.0.0:61438 beunruhigt mich.

Das ist ok, der Port 61438 wird automatisch geöffnet, um den für RTP zu benutzen

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Vielen Dank für Ihre Hilfe!

Ich kann schon einen Erfolg melden!

Heraustelefonieren klappt problemlos, beim Anrufen der öffentlichen MSN's klingelt kein Telefon. Es ist eine Ansage zu hören, die von der Audioqualität so schlecht ist, dass man sie nicht verstehen kann. Wer erzeugt diese Ansage, warum ist die Qualität so schlecht? Könnte man das verbessern?

Noch eine Frage zu meinem vorherigen Punkt 5). An welcher Stelle gibt man die ANI ein?

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Hallo,

 

Kann es sein, dass Sie PCMU statt PCMA als Codec benutzen? Kontrollieren Sie bitte im Netborder Gateway bei Configuration -> PSTN Config -> Physical Configuration -> Karte wählen -> Default PCM law sollte auf "PCMA" stehen.

Danach auch im SnomONE Manager -> Leitungen -> NBE -> Codec Präferenz: auf der linke Seite sollte G711A stehen (G711A = PCMA)

 

>Noch eine Frage zu meinem vorherigen Punkt 5). An welcher Stelle gibt man die ANI ein?

im SnomONE Manager -> Konten -> Nebestelle wählen -> ANI

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Danke für Ihre Rückantwort!

OK, Ihre erste Anregung habe ich kontrolliert. Alles war wie Sie es wollten: PCMA im NBE und SnomONE Leitungen NBE. Leider keine Verbesserung! Beim Anrufen auf die SnomONEbekomme ich eine unverständliche Ansage. Ich glaubte, die Ansage möchte, dass ich die Nummer der anzurufenden Nebenstelle eingebe und habe das getan. Dann kommt wieder eine unverständliche Ansage, aber die Nebenstelle läutet, die Sprachqualität ist unverständlich. Auch das Heraustelefonieren ist wegen der unverständlichen Sprachqualität nicht möglich. Zwar klingt es beim Angerufenen, aber man kann nichts verstehen.

Bei Telefonaten via Sipgate ist die Sprachqualität einwandfrei!

Außerdem höre ich als Anrufer, weder von innen nach außen, noch von außen nach innen ein Freizeichen. Kann man das ändern?

Das logging zu einem Anruf von einem GSM-Handy auf die SnomONE kommt hier:

[5] 2012/03/21 19:59:32: SIP Rx udp:127.0.0.1:5066:

INVITE sip:342410@localhost:5060;transport=udp;user=phone SIP/2.0

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583842-300178843-143ds-632a2131-91369cd8

To: "342410" <sip:342410@localhost:5060>

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

CSeq: 13274419 INVITE

Content-Length: 278

Expires: 179

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

Supported: replaces

Supported: 100rel

User-Agent: Netborder Express Gateway/4.1.6

Content-Type: application/sdp

Date: Wed, 21 Mar 2012 18:59:32 GMT

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bKfe8a6ca4-1dd1-11b2-a128-b8bb90cc0de8

Max-Forwards: 70

 

v=0

o=Sangoma-Tech 1332356372 1332356421 IN IP4 127.0.0.1

s=SIP Call

c=IN IP4 192.168.100.9

t=0 0

m=audio 14022 RTP/AVP 0 8 101 13

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:13 CN/8000

a=ptime:20

a=sendrecv

[8] 2012/03/21 19:59:32: Allocating for call port 14, SIP call id fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

[9] 2012/03/21 19:59:32: UDP(IPv4): Opening socket on 0.0.0.0:56238

[9] 2012/03/21 19:59:32: UDP(IPv4): Opening socket on 0.0.0.0:56239

[9] 2012/03/21 19:59:32: UDP(IPv6): Opening socket on [::]:56238

[9] 2012/03/21 19:59:32: UDP(IPv6): Opening socket on [::]:56239

[5] 2012/03/21 19:59:32: Identify trunk (IP address/port and domain match) 3

[9] 2012/03/21 19:59:32: Resolve 163: aaaa udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 163: a udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 163: udp 127.0.0.1 5066

[5] 2012/03/21 19:59:32: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bKfe8a6ca4-1dd1-11b2-a128-b8bb90cc0de8

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583842-300178843-143ds-632a2131-91369cd8

To: "342410" <sip:342410@localhost:5060>;tag=b0bc908afc

Call-ID: fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

CSeq: 13274419 INVITE

Content-Length: 0

 

[7] 2012/03/21 19:59:32: Set packet length to 20

[6] 2012/03/21 19:59:32: Call-leg 14: Sending RTP for fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil to 192.168.100.9:14022, codec not set yet

[8] 2012/03/21 19:59:32: Incoming call: Request URI sip:342410@localhost:5060;transport=udp;user=phone, To is "342410" <sip:342410@localhost:5060>

[8] 2012/03/21 19:59:32: Call from a trunk 3

[8] 2012/03/21 19:59:32: Trunk NBE@pbx.company.com has country code 49, area code 8458

[9] 2012/03/21 19:59:32: Incoming: formatted From is = "unknown-caller-name" <sip:+4984581796946824@192.168.100.9:5066;user=phone>

[9] 2012/03/21 19:59:32: Incoming: formatted To is = "342410" <sip:+498458342410@localhost:5060;user=phone>

[9] 2012/03/21 19:59:32: Incoming: formatted URI is = sip:+498458342410@pbx.company.com:5060;transport=udp;user=phone

[8] 2012/03/21 19:59:32: Trunk: Check if the call to +498458342410 comes from the cell phone +4984581796946824

[5] 2012/03/21 19:59:32: SIP Rx udp:127.0.0.1:5066:

INVITE sip:342410@localhost:5060;transport=udp;user=phone SIP/2.0

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583487-677973377-142ds-2959c636-91375370

To: "342410" <sip:342410@localhost:5060>

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

CSeq: 4110137 INVITE

Content-Length: 278

Expires: 179

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

Supported: replaces

Supported: 100rel

User-Agent: Netborder Express Gateway/4.1.6

Content-Type: application/sdp

Date: Wed, 21 Mar 2012 18:59:32 GMT

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bKfe8c1b94-1dd1-11b2-b67a-fb7e1c1dfb10

Max-Forwards: 70

 

v=0

o=Sangoma-Tech 1332356372 1332356421 IN IP4 127.0.0.1

s=SIP Call

c=IN IP4 192.168.100.9

t=0 0

m=audio 14024 RTP/AVP 0 8 101 13

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:13 CN/8000

a=ptime:20

a=sendrecv

[8] 2012/03/21 19:59:32: To is "342410" <sip:+498458342410@localhost:5060;user=phone>, user 0, domain 1

[8] 2012/03/21 19:59:32: Allocating for call port 15, SIP call id fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

[8] 2012/03/21 19:59:32: Send call to extension ERE returned 70

[8] 2012/03/21 19:59:32: Looking for EPID 70

[9] 2012/03/21 19:59:32: UDP(IPv4): Opening socket on 0.0.0.0:56756

[5] 2012/03/21 19:59:32: Domain trunk NBE@pbx.company.com sends call to 70 in domain pbx.company.com

[9] 2012/03/21 19:59:32: UDP(IPv4): Opening socket on 0.0.0.0:56757

[9] 2012/03/21 19:59:32: UDP(IPv6): Opening socket on [::]:56756

[9] 2012/03/21 19:59:32: UDP(IPv6): Opening socket on [::]:56757

[8] 2012/03/21 19:59:32: Set the To domain based on To user 70@pbx.company.com

[5] 2012/03/21 19:59:32: Identify trunk (IP address/port and domain match) 3

[8] 2012/03/21 19:59:32: Call state for call object 6: idle

[9] 2012/03/21 19:59:32: Resolve 164: aaaa udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 164: a udp 127.0.0.1 5066

[8] 2012/03/21 19:59:32: Call state for call object 6: connected

[9] 2012/03/21 19:59:32: Resolve 164: udp 127.0.0.1 5066

[8] 2012/03/21 19:59:32: Play audio_moh/noise.wav, caching true

[5] 2012/03/21 19:59:32: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bKfe8c1b94-1dd1-11b2-b67a-fb7e1c1dfb10

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583487-677973377-142ds-2959c636-91375370

To: "342410" <sip:342410@localhost:5060>;tag=8fea595310

Call-ID: fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

CSeq: 4110137 INVITE

Content-Length: 0

 

[5] 2012/03/21 19:59:32: Call port 14: set_codecs for fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil: reached limit on g729

[7] 2012/03/21 19:59:32: Call port 14: set_codecs for fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil codecs "8 9 0 3 2 18", codec_preference count 6

[7] 2012/03/21 19:59:32: Set packet length to 20

[9] 2012/03/21 19:59:32: Call port 14: update_codecs for fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil: adding codec pcma/8000 to available list

[9] 2012/03/21 19:59:32: Call port 14: update_codecs for fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil: Other side does not support codec g722/8000

[9] 2012/03/21 19:59:32: Call port 14: update_codecs for fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil: adding codec pcmu/8000 to available list

[9] 2012/03/21 19:59:32: Call port 14: update_codecs for fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil: Other side does not support codec gsm/8000

[9] 2012/03/21 19:59:32: Call port 14: update_codecs for fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil: Other side does not support codec g726-32/8000

[9] 2012/03/21 19:59:32: Call port 14: update_codecs for fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil: codec_preference size 6, available codecs size 3

[6] 2012/03/21 19:59:32: Call-leg 14: Codec pcma/8000 is chosen for call id fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

[5] 2012/03/21 19:59:32: set codec: codec pcma/8000 is set to call-leg 14

[9] 2012/03/21 19:59:32: Resolve 165: aaaa udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 165: a udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 165: udp 127.0.0.1 5066

[5] 2012/03/21 19:59:32: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 183 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bKfe8a6ca4-1dd1-11b2-a128-b8bb90cc0de8

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583842-300178843-143ds-632a2131-91369cd8

To: "342410" <sip:342410@localhost:5060>;tag=b0bc908afc

Call-ID: fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

CSeq: 13274419 INVITE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 269

 

v=0

o=- 1428913310 1428913310 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 56238 RTP/AVP 8 0 101

a=rtpmap:8 pcma/8000

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

[7] 2012/03/21 19:59:32: Set packet length to 20

[6] 2012/03/21 19:59:32: Call-leg 15: Sending RTP for fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil to 192.168.100.9:14024, codec not set yet

[5] 2012/03/21 19:59:32: SIP Rx udp:127.0.0.1:5066:

PRACK sip:342410@localhost:5060;user=phone;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583842-300178843-143ds-632a2131-91369cd8

To: "342410" <sip:342410@localhost:5060>;tag=b0bc908afc

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

CSeq: 13274420 PRACK

Content-Length: 0

RAck: 1 13274419 INVITE

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bKfe916982-1dd1-11b2-9559-d915b8cdbaa8

 

[9] 2012/03/21 19:59:32: Resolve 166: aaaa udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 166: a udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 166: udp 127.0.0.1 5066

[5] 2012/03/21 19:59:32: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bKfe916982-1dd1-11b2-9559-d915b8cdbaa8

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583842-300178843-143ds-632a2131-91369cd8

To: "342410" <sip:342410@localhost:5060>;tag=b0bc908afc

Call-ID: fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

CSeq: 13274420 PRACK

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[8] 2012/03/21 19:59:32: Incoming call: Request URI sip:342410@localhost:5060;transport=udp;user=phone, To is "342410" <sip:342410@localhost:5060>

[8] 2012/03/21 19:59:32: Call from a trunk 3

[8] 2012/03/21 19:59:32: Trunk NBE@pbx.company.com has country code 49, area code 8458

[9] 2012/03/21 19:59:32: Incoming: formatted From is = "unknown-caller-name" <sip:+4984581796946824@192.168.100.9:5066;user=phone>

[9] 2012/03/21 19:59:32: Incoming: formatted To is = "342410" <sip:+498458342410@localhost:5060;user=phone>

[9] 2012/03/21 19:59:32: Incoming: formatted URI is = sip:+498458342410@pbx.company.com:5060;transport=udp;user=phone

[8] 2012/03/21 19:59:32: Trunk: Check if the call to +498458342410 comes from the cell phone +4984581796946824

[8] 2012/03/21 19:59:32: To is "342410" <sip:+498458342410@localhost:5060;user=phone>, user 0, domain 1

[8] 2012/03/21 19:59:32: Send call to extension ERE returned 70

[8] 2012/03/21 19:59:32: Looking for EPID 70

[5] 2012/03/21 19:59:32: Domain trunk NBE@pbx.company.com sends call to 70 in domain pbx.company.com

[8] 2012/03/21 19:59:32: Set the To domain based on To user 70@pbx.company.com

[8] 2012/03/21 19:59:32: Call state for call object 7: idle

[5] 2012/03/21 19:59:32: Call port 15: set_codecs for fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil: reached limit on g729

[7] 2012/03/21 19:59:32: Call port 15: set_codecs for fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil codecs "8 9 0 3 2 18", codec_preference count 6

[8] 2012/03/21 19:59:32: Call state for call object 7: connected

[8] 2012/03/21 19:59:32: Play audio_moh/noise.wav, caching true

[7] 2012/03/21 19:59:32: Set packet length to 20

[9] 2012/03/21 19:59:32: Call port 15: update_codecs for fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil: adding codec pcma/8000 to available list

[9] 2012/03/21 19:59:32: Call port 15: update_codecs for fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil: Other side does not support codec g722/8000

[9] 2012/03/21 19:59:32: Call port 15: update_codecs for fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil: adding codec pcmu/8000 to available list

[9] 2012/03/21 19:59:32: Call port 15: update_codecs for fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil: Other side does not support codec gsm/8000

[9] 2012/03/21 19:59:32: Call port 15: update_codecs for fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil: Other side does not support codec g726-32/8000

[9] 2012/03/21 19:59:32: Call port 15: update_codecs for fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil: codec_preference size 6, available codecs size 3

[6] 2012/03/21 19:59:32: Call-leg 15: Codec pcma/8000 is chosen for call id fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

[5] 2012/03/21 19:59:32: set codec: codec pcma/8000 is set to call-leg 15

[9] 2012/03/21 19:59:32: Resolve 167: aaaa udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 167: a udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 167: udp 127.0.0.1 5066

[5] 2012/03/21 19:59:32: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 183 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bKfe8c1b94-1dd1-11b2-b67a-fb7e1c1dfb10

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583487-677973377-142ds-2959c636-91375370

To: "342410" <sip:342410@localhost:5060>;tag=8fea595310

Call-ID: fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

CSeq: 4110137 INVITE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 267

 

v=0

o=- 410590064 410590064 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 56756 RTP/AVP 8 0 101

a=rtpmap:8 pcma/8000

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

[6] 2012/03/21 19:59:32: Call-leg 14: Sending RTP for fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil to 127.0.0.1:14022, codec pcma/8000

[5] 2012/03/21 19:59:32: SIP Rx udp:127.0.0.1:5066:

PRACK sip:342410@localhost:5060;user=phone;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583487-677973377-142ds-2959c636-91375370

To: "342410" <sip:342410@localhost:5060>;tag=8fea595310

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

CSeq: 4110138 PRACK

Content-Length: 0

RAck: 1 4110137 INVITE

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bKfe959dae-1dd1-11b2-995f-8e083be75113

 

[9] 2012/03/21 19:59:32: Resolve 168: aaaa udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 168: a udp 127.0.0.1 5066

[9] 2012/03/21 19:59:32: Resolve 168: udp 127.0.0.1 5066

[5] 2012/03/21 19:59:32: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bKfe959dae-1dd1-11b2-995f-8e083be75113

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583487-677973377-142ds-2959c636-91375370

To: "342410" <sip:342410@localhost:5060>;tag=8fea595310

Call-ID: fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

CSeq: 4110138 PRACK

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[6] 2012/03/21 19:59:32: Call-leg 15: Sending RTP for fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil to 127.0.0.1:14024, codec pcma/8000

[8] 2012/03/21 19:59:33: Attendant: Timeout (wait)

[8] 2012/03/21 19:59:33: Play audio_de/aa_welcome_auto.wav space20, caching false

[9] 2012/03/21 19:59:33: Resolve 169: aaaa udp 127.0.0.1 5066

[9] 2012/03/21 19:59:33: Resolve 169: a udp 127.0.0.1 5066

[9] 2012/03/21 19:59:33: Resolve 169: udp 127.0.0.1 5066

[5] 2012/03/21 19:59:33: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bKfe8a6ca4-1dd1-11b2-a128-b8bb90cc0de8

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583842-300178843-143ds-632a2131-91369cd8

To: "342410" <sip:342410@localhost:5060>;tag=b0bc908afc

Call-ID: fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

CSeq: 13274419 INVITE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Type: application/sdp

Content-Length: 269

 

v=0

o=- 1428913310 1428913310 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 56238 RTP/AVP 8 0 101

a=rtpmap:8 pcma/8000

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

[5] 2012/03/21 19:59:33: SIP Rx udp:127.0.0.1:5066:

ACK sip:342410@127.0.0.1:5060;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583842-300178843-143ds-632a2131-91369cd8

To: "342410" <sip:342410@localhost:5060>;tag=b0bc908afc

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

CSeq: 13274419 ACK

Content-Length: 0

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bKff2a471a-1dd1-11b2-8327-ccb406f219b8

 

[8] 2012/03/21 19:59:33: Attendant: Timeout (wait)

[8] 2012/03/21 19:59:33: Play audio_de/aa_welcome_auto.wav space20, caching false

[9] 2012/03/21 19:59:33: Resolve 170: aaaa udp 127.0.0.1 5066

[9] 2012/03/21 19:59:33: Resolve 170: a udp 127.0.0.1 5066

[9] 2012/03/21 19:59:33: Resolve 170: udp 127.0.0.1 5066

[5] 2012/03/21 19:59:33: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bKfe8c1b94-1dd1-11b2-b67a-fb7e1c1dfb10

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583487-677973377-142ds-2959c636-91375370

To: "342410" <sip:342410@localhost:5060>;tag=8fea595310

Call-ID: fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

CSeq: 4110137 INVITE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Type: application/sdp

Content-Length: 267

 

v=0

o=- 410590064 410590064 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 56756 RTP/AVP 8 0 101

a=rtpmap:8 pcma/8000

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

[5] 2012/03/21 19:59:33: SIP Rx udp:127.0.0.1:5066:

ACK sip:342410@127.0.0.1:5060;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583487-677973377-142ds-2959c636-91375370

To: "342410" <sip:342410@localhost:5060>;tag=8fea595310

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

CSeq: 4110137 ACK

Content-Length: 0

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bKff2e6aac-1dd1-11b2-93b8-aff7f6aa80a6

 

[5] 2012/03/21 19:59:33: SIP Rx udp:127.0.0.1:5066:

BYE sip:342410@127.0.0.1:5060;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583487-677973377-142ds-2959c636-91375370

To: "342410" <sip:342410@localhost:5060>;tag=8fea595310

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

CSeq: 4110139 BYE

Content-Length: 0

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bKff3ce168-1dd1-11b2-a047-ab2bb5e52dd1

 

[9] 2012/03/21 19:59:33: Resolve 171: aaaa udp 127.0.0.1 5066

[9] 2012/03/21 19:59:33: Resolve 171: a udp 127.0.0.1 5066

[9] 2012/03/21 19:59:33: Resolve 171: udp 127.0.0.1 5066

[5] 2012/03/21 19:59:33: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bKff3ce168-1dd1-11b2-a047-ab2bb5e52dd1

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583487-677973377-142ds-2959c636-91375370

To: "342410" <sip:342410@localhost:5060>;tag=8fea595310

Call-ID: fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

CSeq: 4110139 BYE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[8] 2012/03/21 19:59:33: Clearing call port 15, SIP call id fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

[8] 2012/03/21 19:59:33: Remove leg 10: call port 15, SIP call id fe8b97c8-1dd1-11b2-8796-af192d93a696@snom.home.nil

[8] 2012/03/21 19:59:33: Hangup: Call 15 not found

[8] 2012/03/21 19:59:37: Last message repeated 2 times

[7] 2012/03/21 19:59:37: Received RFC4733 DTMF on codec 101

[6] 2012/03/21 19:59:37: Received DTMF 4, call type attendant

[6] 2012/03/21 19:59:38: Received DTMF 0, call type attendant

[8] 2012/03/21 19:59:38: Play audio_de/aa_say_extension.wav audio_de/bi_4.wav audio_de/bi_0.wav, caching false

[8] 2012/03/21 19:59:38: Attendant: Ignoring the DTMF 0 in the state say_name

[8] 2012/03/21 19:59:40: Call state for call object 6: alerting

[8] 2012/03/21 19:59:40: Play audio_moh/noise.wav, caching true

[8] 2012/03/21 19:59:40: Allocating for call port 16, SIP call id f85cc1c4@pbx

[9] 2012/03/21 19:59:40: UDP(IPv4): Opening socket on 0.0.0.0:54668

[9] 2012/03/21 19:59:40: UDP(IPv4): Opening socket on 0.0.0.0:54669

[9] 2012/03/21 19:59:40: UDP(IPv6): Opening socket on [::]:54668

[9] 2012/03/21 19:59:40: UDP(IPv6): Opening socket on [::]:54669

[7] 2012/03/21 19:59:40: Call port 16: set_codecs for f85cc1c4@pbx codecs "", codec_preference count 6

[9] 2012/03/21 19:59:40: Using outbound proxy sip:192.168.100.28:4411;transport=tls because of flow-label

[9] 2012/03/21 19:59:40: Call port 16: update_codecs for f85cc1c4@pbx: adding codec pcma/8000 to available list

[9] 2012/03/21 19:59:40: Call port 16: update_codecs for f85cc1c4@pbx: adding codec pcmu/8000 to available list

[9] 2012/03/21 19:59:40: Call port 16: update_codecs for f85cc1c4@pbx: adding codec g722/8000 to available list

[9] 2012/03/21 19:59:40: Call port 16: update_codecs for f85cc1c4@pbx: adding codec g726-32/8000 to available list

[9] 2012/03/21 19:59:40: Call port 16: update_codecs for f85cc1c4@pbx: adding codec gsm/8000 to available list

[9] 2012/03/21 19:59:40: Call port 16: update_codecs for f85cc1c4@pbx: codec_preference size 6, available codecs size 6

[5] 2012/03/21 19:59:40: SIP Tx tls:192.168.100.28:4411:

INVITE sip:40@192.168.100.28:4411;transport=tls;line=0ii3wcy6 SIP/2.0

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-a026be46b998a307153eb720eba0adb0;rport

From: "unknown-caller-name" <sip:1796946824@pbx.company.com:5066;user=phone>;tag=648621777

To: "Walter Hackner" <sip:40@pbx.company.com>

Call-ID: f85cc1c4@pbx

CSeq: 23087 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.100.9:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Alert-Info: <http://127.0.0.1/Bellcore-dr3>

Content-Type: application/sdp

Content-Length: 421

 

v=0

o=- 250173557 250173557 IN IP4 192.168.100.9

s=-

c=IN IP4 192.168.100.9

t=0 0

m=audio 54668 RTP/AVP 8 0 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:OWhHUmbPMtfHOqAbN4biVk+YAmuJz9ta/ihgiz6N

a=rtpmap:8 pcma/8000

a=rtpmap:0 pcmu/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

[5] 2012/03/21 19:59:40: SIP Rx tls:192.168.100.28:4411:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-a026be46b998a307153eb720eba0adb0;rport=5061

From: "unknown-caller-name" <sip:1796946824@pbx.company.com:5066;user=phone>;tag=648621777

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=hjv9w9r5vx

Call-ID: f85cc1c4@pbx

CSeq: 23087 INVITE

Contact: <sip:40@192.168.100.28:4411;transport=tls;line=0ii3wcy6>;reg-id=1

Require: 100rel

RSeq: 1

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

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

Content-Length: 0

 

[5] 2012/03/21 19:59:40: SIP Tx tls:192.168.100.28:4411:

PRACK sip:40@192.168.100.28:4411;transport=tls;line=0ii3wcy6 SIP/2.0

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-29e453c84c18021e4d52a3b65ac60d86;rport

From: "unknown-caller-name" <sip:1796946824@pbx.company.com:5066;user=phone>;tag=648621777

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=hjv9w9r5vx

Call-ID: f85cc1c4@pbx

CSeq: 23088 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.100.9:5061;transport=tls>

RAck: 1 23087 INVITE

Content-Length: 0

 

[8] 2012/03/21 19:59:40: Play audio_de/ringback.wav, caching true

[5] 2012/03/21 19:59:40: SIP Rx tls:192.168.100.28:4411:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-29e453c84c18021e4d52a3b65ac60d86;rport=5061

From: "unknown-caller-name" <sip:1796946824@pbx.company.com:5066;user=phone>;tag=648621777

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=hjv9w9r5vx

Call-ID: f85cc1c4@pbx

CSeq: 23088 PRACK

Contact: <sip:40@192.168.100.28:4411;transport=tls;line=0ii3wcy6>;reg-id=1

Content-Length: 0

 

[7] 2012/03/21 19:59:40: Call f85cc1c4@pbx: Clear last request

[5] 2012/03/21 19:59:44: SIP Rx udp:127.0.0.1:5066:

BYE sip:342410@127.0.0.1:5060;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583842-300178843-143ds-632a2131-91369cd8

To: "342410" <sip:342410@localhost:5060>;tag=b0bc908afc

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

CSeq: 13274421 BYE

Content-Length: 0

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bK05803f84-1dd2-11b2-bf7d-ef2af8b25909

 

[9] 2012/03/21 19:59:44: Resolve 174: aaaa udp 127.0.0.1 5066

[9] 2012/03/21 19:59:44: Resolve 174: a udp 127.0.0.1 5066

[9] 2012/03/21 19:59:44: Resolve 174: udp 127.0.0.1 5066

[5] 2012/03/21 19:59:44: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK05803f84-1dd2-11b2-bf7d-ef2af8b25909

From: "unknown-caller-name" <sip:1796946824@192.168.100.9:5066>;tag=pxip-callid-1332356372-583842-300178843-143ds-632a2131-91369cd8

To: "342410" <sip:342410@localhost:5060>;tag=b0bc908afc

Call-ID: fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

CSeq: 13274421 BYE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[8] 2012/03/21 19:59:44: Clearing call port 14, SIP call id fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

[5] 2012/03/21 19:59:44: SIP Tx tls:192.168.100.28:4411:

CANCEL sip:40@192.168.100.28:4411;transport=tls;line=0ii3wcy6 SIP/2.0

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-a026be46b998a307153eb720eba0adb0;rport

From: "unknown-caller-name" <sip:1796946824@pbx.company.com:5066;user=phone>;tag=648621777

To: "Walter Hackner" <sip:40@pbx.company.com>

Call-ID: f85cc1c4@pbx

CSeq: 23087 CANCEL

Max-Forwards: 70

Content-Length: 0

 

[8] 2012/03/21 19:59:44: Remove leg 9: call port 14, SIP call id fe89d078-1dd1-11b2-94d0-985e00e3764e@snom.home.nil

[8] 2012/03/21 19:59:44: Hangup: Call 14 not found

[8] 2012/03/21 19:59:44: Last message repeated 2 times

[5] 2012/03/21 19:59:44: SIP Rx tls:192.168.100.28:4411:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-a026be46b998a307153eb720eba0adb0;rport=5061

From: "unknown-caller-name" <sip:1796946824@pbx.company.com:5066;user=phone>;tag=648621777

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=hjv9w9r5vx

Call-ID: f85cc1c4@pbx

CSeq: 23087 CANCEL

Content-Length: 0

 

[7] 2012/03/21 19:59:44: Call f85cc1c4@pbx: Clear last request

[5] 2012/03/21 19:59:44: SIP Rx tls:192.168.100.28:4411:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-a026be46b998a307153eb720eba0adb0;rport=5061

From: "unknown-caller-name" <sip:1796946824@pbx.company.com:5066;user=phone>;tag=648621777

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=hjv9w9r5vx

Call-ID: f85cc1c4@pbx

CSeq: 23087 INVITE

Contact: <sip:40@192.168.100.28:4411;transport=tls;line=0ii3wcy6>;reg-id=1

Content-Length: 0

 

[7] 2012/03/21 19:59:44: Call f85cc1c4@pbx: Clear last INVITE

[5] 2012/03/21 19:59:44: SIP Tx tls:192.168.100.28:4411:

ACK sip:40@192.168.100.28:4411;transport=tls;line=0ii3wcy6 SIP/2.0

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-a026be46b998a307153eb720eba0adb0;rport

From: "unknown-caller-name" <sip:1796946824@pbx.company.com:5066;user=phone>;tag=648621777

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=hjv9w9r5vx

Call-ID: f85cc1c4@pbx

CSeq: 23087 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.100.9:5061;transport=tls>

Content-Length: 0

 

[5] 2012/03/21 19:59:44: INVITE Response 487 Request Terminated: Terminate f85cc1c4@pbx

[8] 2012/03/21 19:59:44: Clearing call port 16, SIP call id f85cc1c4@pbx

[8] 2012/03/21 19:59:44: Remove leg 11: call port 16, SIP call id f85cc1c4@pbx

 

 

Zu meiner Frage 5) aus dem vorletzten Posting: Wenn ich will, dass eine bestimmte MSN bei ausgehenden Telefonaten gesendet werden soll, gebe ich die MSN bei ANI des zuständigen Kontos ein? Wie funktioniert die Sache, wenn ich einen 10er-Block MSN's von der Telekom bekommen habe und der SnomONE sagen möchte, bei welchen Nummern sie drangehen soll und an welche Nebenstelle weitervermittelt werden soll?

 

Vielen Dank für Ihre Mühe!

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Hallo,

 

Sind Sie sicher, dass nur pcma für den Trunk in der Codecliste steht? Ich sehe beide pcma und pcmu. Bitte lassen Sie nur pcma, kein pcmu.

 

>Wenn ich will, dass eine bestimmte MSN bei ausgehenden Telefonaten gesendet werden soll, gebe ich die MSN bei ANI des zuständigen Kontos ein?

genau

 

>Wie funktioniert die Sache, wenn ich einen 10er-Block MSN's von der Telekom bekommen habe und der SnomONE sagen möchte, bei welchen Nummern sie drangehen soll und an welche Nebenstelle weitervermittelt werden soll?

Dafür muss man als Ziel Konto auch ein komplexer String benutzen. zB wenn Ihre MSNs 2 Zahlen haben, können Sie die snom ONE Nebestelle gleich nummerieren, und dann als Ziel Konto:

!([0-9]{2}$)!\1!

(mehr Info hier: http://wiki.snomone.com/index.php?title=Inbounds_Calls#How_the_System_Routes_a_Call_to_the_Proper_Extension)

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Hallo,

danke für Ihre Antwort! Leider komme ich erst jetzt dazu, Ihre Anmerkungen zu kontrollieren:

1) bei Leitungen->Ändern finde ich beim trunk 'NBE' in der unteren Hälfte den Punkt 'Media/Audio: Codec Präferenz:'. Dort sind 2 Fenster. Im rechten sind mehrere Codecs zu sehen, das linke ist leer. Ich habe es mit G.711A gefüllt. Im NBE Gateway habe ich überprüft bei 'Default PCM law' steht pcma!

Ist das genug und richtig?

2) Was ist ein komplexer String zur Eintragung bei Ziel Konto? Ist das '(mehr Info hier: http://wiki.snomone....roper_Extension)' damit gemeint?

Danke für Ihre Hilfe!

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Hallo,

nun habe ich alle Anweisungen umgesetzt!

Am 6.4. um ca. 11:43 habe das Problem durch einen Anruf von extern(=Handy) zur SnomOne und einen Anruf von der SnomOne nach extern reproduziert!

Hier das Protokoll:

 

[5] 2012/04/06 11:43:28: SIP Rx tls:192.168.100.28:3857:

SUBSCRIBE sip:192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-6dxtxjwh6mw4;rport

From: <sip:40@pbx.company.com>;tag=xh6zdvdigt

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=0ceb9e1a8b

Call-ID: 153d3a3cc317-b7c03ro5zct8

CSeq: 14819 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

Event: as-feature-event

User-Agent: snom870/8.4.18

Proxy-Require: buttons

Expires: 3600

Content-Type: application/x-as-feature-event+xml

Content-Length: 0

 

[8] 2012/04/06 11:43:28: Packet authenticated by transport layer

[5] 2012/04/06 11:43:28: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-6dxtxjwh6mw4;rport=3857

From: <sip:40@pbx.company.com>;tag=xh6zdvdigt

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=0ceb9e1a8b

Call-ID: 153d3a3cc317-b7c03ro5zct8

CSeq: 14819 SUBSCRIBE

Contact: <sip:192.168.100.9:5061;transport=tls>

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Expires: 179

Content-Length: 0

 

[9] 2012/04/06 11:43:44: Remote site 192.168.100.37 closed the connection

[9] 2012/04/06 11:43:54: Last message repeated 2 times

[5] 2012/04/06 11:43:54: SIP Rx udp:127.0.0.1:5066:

INVITE sip:342410@localhost:5060;transport=udp;user=phone SIP/2.0

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-634854-1841585795-105ds-1c69fb81-faaca7f0

To: "342410" <sip:342410@localhost:5060>

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

CSeq: 8385425 INVITE

Content-Length: 278

Expires: 179

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

Supported: replaces

Supported: 100rel

User-Agent: Netborder Express Gateway/4.1.6

Content-Type: application/sdp

Date: Fri, 06 Apr 2012 09:43:54 GMT

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bK06330e88-1dd2-11b2-9a51-ad7fdc099585

Max-Forwards: 70

 

v=0

o=Sangoma-Tech 1333705434 1333705483 IN IP4 127.0.0.1

s=SIP Call

c=IN IP4 192.168.100.9

t=0 0

m=audio 14000 RTP/AVP 0 8 101 13

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:13 CN/8000

a=ptime:20

a=sendrecv

[8] 2012/04/06 11:43:54: Allocating for call port 0, SIP call id 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

[9] 2012/04/06 11:43:54: UDP(IPv4): Opening socket on 0.0.0.0:62038

[9] 2012/04/06 11:43:54: UDP(IPv4): Opening socket on 0.0.0.0:62039

[9] 2012/04/06 11:43:54: UDP(IPv6): Opening socket on [::]:62038

[9] 2012/04/06 11:43:54: UDP(IPv6): Opening socket on [::]:62039

[5] 2012/04/06 11:43:54: Identify trunk (IP address/port and domain match) 3

[9] 2012/04/06 11:43:54: Resolve 19776: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19776: a udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19776: udp 127.0.0.1 5066

[5] 2012/04/06 11:43:54: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK06330e88-1dd2-11b2-9a51-ad7fdc099585

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-634854-1841585795-105ds-1c69fb81-faaca7f0

To: "342410" <sip:342410@localhost:5060>;tag=d8365b0f5d

Call-ID: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

CSeq: 8385425 INVITE

Content-Length: 0

 

[7] 2012/04/06 11:43:54: Set packet length to 20

[6] 2012/04/06 11:43:54: Call-leg 0: Sending RTP for 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil to 192.168.100.9:14000, codec not set yet

[8] 2012/04/06 11:43:54: Incoming call: Request URI sip:342410@localhost:5060;transport=udp;user=phone, To is "342410" <sip:342410@localhost:5060>

[8] 2012/04/06 11:43:54: Call from a trunk 3

[8] 2012/04/06 11:43:54: Trunk NBE@pbx.company.com has country code 49, area code 8458

[9] 2012/04/06 11:43:54: Incoming: formatted From is = "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066;user=phone>

[9] 2012/04/06 11:43:54: Incoming: formatted To is = "342410" <sip:+498458342410@localhost:5060;user=phone>

[9] 2012/04/06 11:43:54: Incoming: formatted URI is = sip:+498458342410@pbx.company.com:5060;transport=udp;user=phone

[8] 2012/04/06 11:43:54: Trunk: Check if the call to +498458342410 comes from the cell phone unknown-ani

[5] 2012/04/06 11:43:54: SIP Rx udp:127.0.0.1:5066:

INVITE sip:342410@localhost:5060;transport=udp;user=phone SIP/2.0

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-627575-561717988-103ds-1e23ef54-faadf3a8

To: "342410" <sip:342410@localhost:5060>

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

CSeq: 4235573 INVITE

Content-Length: 278

Expires: 179

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

Supported: replaces

Supported: 100rel

User-Agent: Netborder Express Gateway/4.1.6

Content-Type: application/sdp

Date: Fri, 06 Apr 2012 09:43:54 GMT

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bK0634c7e6-1dd2-11b2-94b6-d5a0caeb29fb

Max-Forwards: 70

 

v=0

o=Sangoma-Tech 1333705434 1333705483 IN IP4 127.0.0.1

s=SIP Call

c=IN IP4 192.168.100.9

t=0 0

m=audio 14002 RTP/AVP 0 8 101 13

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:13 CN/8000

a=ptime:20

a=sendrecv

[8] 2012/04/06 11:43:54: To is "342410" <sip:+498458342410@localhost:5060;user=phone>, user 0, domain 1

[8] 2012/04/06 11:43:54: Send call to extension ERE returned 70

[8] 2012/04/06 11:43:54: Allocating for call port 1, SIP call id 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

[8] 2012/04/06 11:43:54: Looking for EPID 70

[9] 2012/04/06 11:43:54: UDP(IPv4): Opening socket on 0.0.0.0:56728

[9] 2012/04/06 11:43:54: UDP(IPv4): Opening socket on 0.0.0.0:56729

[5] 2012/04/06 11:43:54: Domain trunk NBE@pbx.company.com sends call to 70 in domain pbx.company.com

[9] 2012/04/06 11:43:54: UDP(IPv6): Opening socket on [::]:56728

[9] 2012/04/06 11:43:54: UDP(IPv6): Opening socket on [::]:56729

[8] 2012/04/06 11:43:54: Set the To domain based on To user 70@pbx.company.com

[5] 2012/04/06 11:43:54: Identify trunk (IP address/port and domain match) 3

[8] 2012/04/06 11:43:54: Call state for call object 1: idle

[9] 2012/04/06 11:43:54: Resolve 19777: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19777: a udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19777: udp 127.0.0.1 5066

[5] 2012/04/06 11:43:54: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK0634c7e6-1dd2-11b2-94b6-d5a0caeb29fb

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-627575-561717988-103ds-1e23ef54-faadf3a8

To: "342410" <sip:342410@localhost:5060>;tag=04d9de8bef

Call-ID: 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

CSeq: 4235573 INVITE

Content-Length: 0

 

[7] 2012/04/06 11:43:54: Call port 0: set_codecs for 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil codecs "8", codec_preference count 2

[8] 2012/04/06 11:43:54: Call state for call object 1: connected

[8] 2012/04/06 11:43:54: Play audio_moh/noise.wav, caching true

[7] 2012/04/06 11:43:54: Set packet length to 20

[9] 2012/04/06 11:43:54: Call port 0: update_codecs for 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil: adding codec pcma/8000 to available list

[9] 2012/04/06 11:43:54: Call port 0: update_codecs for 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil: codec_preference size 2, available codecs size 2

[6] 2012/04/06 11:43:54: Call-leg 0: Codec pcma/8000 is chosen for call id 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

[5] 2012/04/06 11:43:54: set codec: codec pcma/8000 is set to call-leg 0

[9] 2012/04/06 11:43:54: Resolve 19778: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19778: a udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19778: udp 127.0.0.1 5066

[5] 2012/04/06 11:43:54: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 183 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK06330e88-1dd2-11b2-9a51-ad7fdc099585

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-634854-1841585795-105ds-1c69fb81-faaca7f0

To: "342410" <sip:342410@localhost:5060>;tag=d8365b0f5d

Call-ID: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

CSeq: 8385425 INVITE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 245

 

v=0

o=- 1623961317 1623961317 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 62038 RTP/AVP 8 101

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

[7] 2012/04/06 11:43:54: Set packet length to 20

[6] 2012/04/06 11:43:54: Call-leg 1: Sending RTP for 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil to 192.168.100.9:14002, codec not set yet

[8] 2012/04/06 11:43:54: Incoming call: Request URI sip:342410@localhost:5060;transport=udp;user=phone, To is "342410" <sip:342410@localhost:5060>

[8] 2012/04/06 11:43:54: Call from a trunk 3

[8] 2012/04/06 11:43:54: Trunk NBE@pbx.company.com has country code 49, area code 8458

[9] 2012/04/06 11:43:54: Incoming: formatted From is = "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066;user=phone>

[9] 2012/04/06 11:43:54: Incoming: formatted To is = "342410" <sip:+498458342410@localhost:5060;user=phone>

[9] 2012/04/06 11:43:54: Incoming: formatted URI is = sip:+498458342410@pbx.company.com:5060;transport=udp;user=phone

[8] 2012/04/06 11:43:54: Trunk: Check if the call to +498458342410 comes from the cell phone unknown-ani

[8] 2012/04/06 11:43:54: To is "342410" <sip:+498458342410@localhost:5060;user=phone>, user 0, domain 1

[8] 2012/04/06 11:43:54: Send call to extension ERE returned 70

[8] 2012/04/06 11:43:54: Looking for EPID 70

[5] 2012/04/06 11:43:54: Domain trunk NBE@pbx.company.com sends call to 70 in domain pbx.company.com

[8] 2012/04/06 11:43:54: Set the To domain based on To user 70@pbx.company.com

[8] 2012/04/06 11:43:54: Call state for call object 2: idle

[7] 2012/04/06 11:43:54: Call port 1: set_codecs for 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil codecs "8", codec_preference count 2

[8] 2012/04/06 11:43:54: Call state for call object 2: connected

[8] 2012/04/06 11:43:54: Play audio_moh/noise.wav, caching true

[7] 2012/04/06 11:43:54: Set packet length to 20

[9] 2012/04/06 11:43:54: Call port 1: update_codecs for 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil: adding codec pcma/8000 to available list

[9] 2012/04/06 11:43:54: Call port 1: update_codecs for 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil: codec_preference size 2, available codecs size 2

[6] 2012/04/06 11:43:54: Call-leg 1: Codec pcma/8000 is chosen for call id 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

[5] 2012/04/06 11:43:54: set codec: codec pcma/8000 is set to call-leg 1

[9] 2012/04/06 11:43:54: Resolve 19779: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19779: a udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19779: udp 127.0.0.1 5066

[5] 2012/04/06 11:43:54: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 183 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK0634c7e6-1dd2-11b2-94b6-d5a0caeb29fb

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-627575-561717988-103ds-1e23ef54-faadf3a8

To: "342410" <sip:342410@localhost:5060>;tag=04d9de8bef

Call-ID: 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

CSeq: 4235573 INVITE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 243

 

v=0

o=- 260976677 260976677 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 56728 RTP/AVP 8 101

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

[5] 2012/04/06 11:43:54: SIP Rx udp:127.0.0.1:5066:

PRACK sip:342410@localhost:5060;user=phone;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-634854-1841585795-105ds-1c69fb81-faaca7f0

To: "342410" <sip:342410@localhost:5060>;tag=d8365b0f5d

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

CSeq: 8385426 PRACK

Content-Length: 0

RAck: 1 8385425 INVITE

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bK06399884-1dd2-11b2-9d2b-f0a9a7f3eba1

 

[9] 2012/04/06 11:43:54: Resolve 19780: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19780: a udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19780: udp 127.0.0.1 5066

[5] 2012/04/06 11:43:54: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK06399884-1dd2-11b2-9d2b-f0a9a7f3eba1

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-634854-1841585795-105ds-1c69fb81-faaca7f0

To: "342410" <sip:342410@localhost:5060>;tag=d8365b0f5d

Call-ID: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

CSeq: 8385426 PRACK

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[5] 2012/04/06 11:43:54: SIP Rx udp:127.0.0.1:5066:

PRACK sip:342410@localhost:5060;user=phone;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-627575-561717988-103ds-1e23ef54-faadf3a8

To: "342410" <sip:342410@localhost:5060>;tag=04d9de8bef

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

CSeq: 4235574 PRACK

Content-Length: 0

RAck: 1 4235573 INVITE

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bK063ce03e-1dd2-11b2-a458-a32ee3bab47e

 

[9] 2012/04/06 11:43:54: Resolve 19781: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19781: a udp 127.0.0.1 5066

[9] 2012/04/06 11:43:54: Resolve 19781: udp 127.0.0.1 5066

[5] 2012/04/06 11:43:54: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK063ce03e-1dd2-11b2-a458-a32ee3bab47e

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-627575-561717988-103ds-1e23ef54-faadf3a8

To: "342410" <sip:342410@localhost:5060>;tag=04d9de8bef

Call-ID: 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

CSeq: 4235574 PRACK

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[6] 2012/04/06 11:43:54: Call-leg 0: Sending RTP for 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil to 127.0.0.1:14000, codec pcma/8000

[6] 2012/04/06 11:43:54: Call-leg 1: Sending RTP for 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil to 127.0.0.1:14002, codec pcma/8000

[8] 2012/04/06 11:43:55: Attendant: Timeout (wait)

[8] 2012/04/06 11:43:55: Play audio_de/aa_welcome_auto.wav space20, caching false

[9] 2012/04/06 11:43:55: Resolve 19782: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:43:55: Resolve 19782: a udp 127.0.0.1 5066

[9] 2012/04/06 11:43:55: Resolve 19782: udp 127.0.0.1 5066

[5] 2012/04/06 11:43:55: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK06330e88-1dd2-11b2-9a51-ad7fdc099585

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-634854-1841585795-105ds-1c69fb81-faaca7f0

To: "342410" <sip:342410@localhost:5060>;tag=d8365b0f5d

Call-ID: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

CSeq: 8385425 INVITE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Type: application/sdp

Content-Length: 245

 

v=0

o=- 1623961317 1623961317 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 62038 RTP/AVP 8 101

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

[5] 2012/04/06 11:43:55: SIP Rx udp:127.0.0.1:5066:

ACK sip:342410@127.0.0.1:5060;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-634854-1841585795-105ds-1c69fb81-faaca7f0

To: "342410" <sip:342410@localhost:5060>;tag=d8365b0f5d

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

CSeq: 8385425 ACK

Content-Length: 0

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bK06d18978-1dd2-11b2-91a2-e3475e02c67d

 

[8] 2012/04/06 11:43:55: Attendant: Timeout (wait)

[8] 2012/04/06 11:43:55: Play audio_de/aa_welcome_auto.wav space20, caching false

[9] 2012/04/06 11:43:55: Resolve 19783: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:43:55: Resolve 19783: a udp 127.0.0.1 5066

[9] 2012/04/06 11:43:55: Resolve 19783: udp 127.0.0.1 5066

[5] 2012/04/06 11:43:55: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK0634c7e6-1dd2-11b2-94b6-d5a0caeb29fb

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-627575-561717988-103ds-1e23ef54-faadf3a8

To: "342410" <sip:342410@localhost:5060>;tag=04d9de8bef

Call-ID: 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

CSeq: 4235573 INVITE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Type: application/sdp

Content-Length: 243

 

v=0

o=- 260976677 260976677 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 56728 RTP/AVP 8 101

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

[5] 2012/04/06 11:43:55: SIP Rx udp:127.0.0.1:5066:

ACK sip:342410@127.0.0.1:5060;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-627575-561717988-103ds-1e23ef54-faadf3a8

To: "342410" <sip:342410@localhost:5060>;tag=04d9de8bef

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

CSeq: 4235573 ACK

Content-Length: 0

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bK06d38c28-1dd2-11b2-99e5-b6f85fdc41ba

 

[5] 2012/04/06 11:43:55: SIP Rx udp:127.0.0.1:5066:

BYE sip:342410@127.0.0.1:5060;transport=udp SIP/2.0

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-627575-561717988-103ds-1e23ef54-faadf3a8

To: "342410" <sip:342410@localhost:5060>;tag=04d9de8bef

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Call-ID: 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

CSeq: 4235575 BYE

Content-Length: 0

Max-Forwards: 70

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bK06e38cae-1dd2-11b2-ab09-b6e2e4e66448

 

[9] 2012/04/06 11:43:55: Resolve 19784: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:43:55: Resolve 19784: a udp 127.0.0.1 5066

[9] 2012/04/06 11:43:55: Resolve 19784: udp 127.0.0.1 5066

[5] 2012/04/06 11:43:55: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK06e38cae-1dd2-11b2-ab09-b6e2e4e66448

From: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-627575-561717988-103ds-1e23ef54-faadf3a8

To: "342410" <sip:342410@localhost:5060>;tag=04d9de8bef

Call-ID: 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

CSeq: 4235575 BYE

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[8] 2012/04/06 11:43:55: Clearing call port 1, SIP call id 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

[8] 2012/04/06 11:43:55: Remove leg 2: call port 1, SIP call id 063427e6-1dd2-11b2-832e-a4784f8eda39@snom.home.nil

[8] 2012/04/06 11:43:55: Hangup: Call 1 not found

[8] 2012/04/06 11:44:00: Last message repeated 2 times

[8] 2012/04/06 11:44:00: Play audio_de/aa_enter_extension_number.wav space20, caching false

[5] 2012/04/06 11:44:02: SIP Rx tls:192.168.100.28:3857:

REGISTER sip:pbx.company.com SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-oaf6etxv33tk;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=y6mzqc2syx

To: "Walter Hackner" <sip:40@pbx.company.com>

Call-ID: 3e70263c147b-u4cbzoqqc1vt

CSeq: 32599 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:2dfbe0a8-1848-4d08-9261-4b8e55184131>"

User-Agent: snom870/8.4.18

Allow-Events: dialog

X-Real-IP: 192.168.100.28

Supported: path, gruu

WWW-Contact: <http://192.168.100.28:80>

WWW-Contact: <https://192.168.100.28:443>

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

[8] 2012/04/06 11:44:02: Packet authenticated by transport layer

[5] 2012/04/06 11:44:02: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-oaf6etxv33tk;rport=3857

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=y6mzqc2syx

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=f11c9ea38f

Call-ID: 3e70263c147b-u4cbzoqqc1vt

CSeq: 32599 REGISTER

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;expires=181

Supported: path

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[8] 2012/04/06 11:44:04: Play audio_de/aa_enter_extension_number.wav space20, caching false

[8] 2012/04/06 11:44:07: Trunk 2: Preparing for re-registration

[8] 2012/04/06 11:44:07: Trunk 2: sending discover message for sipgate.de

[9] 2012/04/06 11:44:07: Resolve 19786: url sip:sipgate.de

[9] 2012/04/06 11:44:07: Resolve 19786: naptr sipgate.de

[9] 2012/04/06 11:44:07: Resolve 19786: srv tls _sips._tcp.sipgate.de

[9] 2012/04/06 11:44:07: Resolve 19786: srv tcp _sip._tcp.sipgate.de

[9] 2012/04/06 11:44:07: Resolve 19786: srv udp _sip._udp.sipgate.de

[9] 2012/04/06 11:44:07: Resolve 19786: a udp sipgate.de 5060

[9] 2012/04/06 11:44:07: Resolve 19786: udp 217.10.79.9 5060

[9] 2012/04/06 11:44:07: Resolve 19786: aaaa udp sipgate.de 5060

[9] 2012/04/06 11:44:07: Last message repeated 2 times

[9] 2012/04/06 11:44:07: Resolve 19786: a udp sipgate.de 5060

[9] 2012/04/06 11:44:07: Resolve 19786: udp 217.10.79.9 5060

[8] 2012/04/06 11:44:07: Trunk 2: Received reply for discover method

[8] 2012/04/06 11:44:07: Trunk 2 (sipgate) is associated with the following addresses: udp:217.10.79.9:5060

[5] 2012/04/06 11:44:07: Trunk sipgate is disabled. Not sending registration

[8] 2012/04/06 11:44:08: Play audio_de/aa_enter_extension_number.wav space20, caching false

[8] 2012/04/06 11:44:29: Last message repeated 5 times

[7] 2012/04/06 11:44:29: Received RFC4733 DTMF on codec 101

[6] 2012/04/06 11:44:29: Received DTMF 4, call type attendant

[6] 2012/04/06 11:44:30: Received DTMF 0, call type attendant

[8] 2012/04/06 11:44:30: Play audio_de/aa_say_extension.wav audio_de/bi_4.wav audio_de/bi_0.wav, caching false

[8] 2012/04/06 11:44:30: Attendant: Ignoring the DTMF 0 in the state say_name

[8] 2012/04/06 11:44:32: Call state for call object 1: alerting

[8] 2012/04/06 11:44:32: Play audio_moh/noise.wav, caching true

[8] 2012/04/06 11:44:32: Allocating for call port 2, SIP call id 68e01a15@pbx

[9] 2012/04/06 11:44:32: UDP(IPv4): Opening socket on 0.0.0.0:63094

[9] 2012/04/06 11:44:32: UDP(IPv4): Opening socket on 0.0.0.0:63095

[9] 2012/04/06 11:44:32: UDP(IPv6): Opening socket on [::]:63094

[9] 2012/04/06 11:44:32: UDP(IPv6): Opening socket on [::]:63095

[7] 2012/04/06 11:44:32: Call port 2: set_codecs for 68e01a15@pbx codecs "", codec_preference count 6

[9] 2012/04/06 11:44:32: Using outbound proxy sip:192.168.100.28:3857;transport=tls because of flow-label

[9] 2012/04/06 11:44:32: Call port 2: update_codecs for 68e01a15@pbx: adding codec pcma/8000 to available list

[9] 2012/04/06 11:44:32: Call port 2: update_codecs for 68e01a15@pbx: adding codec pcmu/8000 to available list

[9] 2012/04/06 11:44:32: Call port 2: update_codecs for 68e01a15@pbx: adding codec g722/8000 to available list

[9] 2012/04/06 11:44:32: Call port 2: update_codecs for 68e01a15@pbx: adding codec g726-32/8000 to available list

[9] 2012/04/06 11:44:32: Call port 2: update_codecs for 68e01a15@pbx: adding codec gsm/8000 to available list

[9] 2012/04/06 11:44:32: Call port 2: update_codecs for 68e01a15@pbx: codec_preference size 6, available codecs size 6

[5] 2012/04/06 11:44:32: SIP Tx tls:192.168.100.28:3857:

INVITE sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6 SIP/2.0

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-d3c7b758d1f7eb84428823e1385cecce;rport

From: "unknown-caller-name" <sip:unknown-ani@pbx.company.com:5066;user=phone>;tag=477149378

To: "Walter Hackner" <sip:40@pbx.company.com>

Call-ID: 68e01a15@pbx

CSeq: 14901 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.100.9:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Alert-Info: <http://127.0.0.1/Bellcore-dr3>

Content-Type: application/sdp

Content-Length: 423

 

v=0

o=- 1341584049 1341584049 IN IP4 192.168.100.9

s=-

c=IN IP4 192.168.100.9

t=0 0

m=audio 63094 RTP/AVP 8 0 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:lDOEzMp8OdL0RljtdhTVPJwTzF19DRI/QqdsuOj5

a=rtpmap:8 pcma/8000

a=rtpmap:0 pcmu/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

[5] 2012/04/06 11:44:32: SIP Rx tls:192.168.100.28:3857:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-d3c7b758d1f7eb84428823e1385cecce;rport=5061

From: "unknown-caller-name" <sip:unknown-ani@pbx.company.com:5066;user=phone>;tag=477149378

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=ecutacjb4m

Call-ID: 68e01a15@pbx

CSeq: 14901 INVITE

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

Require: 100rel

RSeq: 1

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

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

Content-Length: 0

 

[5] 2012/04/06 11:44:32: SIP Tx tls:192.168.100.28:3857:

PRACK sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6 SIP/2.0

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-10d94e15402b1ff4dcc71325a4da7221;rport

From: "unknown-caller-name" <sip:unknown-ani@pbx.company.com:5066;user=phone>;tag=477149378

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=ecutacjb4m

Call-ID: 68e01a15@pbx

CSeq: 14902 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.100.9:5061;transport=tls>

RAck: 1 14901 INVITE

Content-Length: 0

 

[8] 2012/04/06 11:44:32: Play audio_de/ringback.wav, caching true

[5] 2012/04/06 11:44:32: SIP Rx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-10d94e15402b1ff4dcc71325a4da7221;rport=5061

From: "unknown-caller-name" <sip:unknown-ani@pbx.company.com:5066;user=phone>;tag=477149378

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=ecutacjb4m

Call-ID: 68e01a15@pbx

CSeq: 14902 PRACK

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

Content-Length: 0

 

[7] 2012/04/06 11:44:32: Call 68e01a15@pbx: Clear last request

[5] 2012/04/06 11:44:36: SIP Rx tls:192.168.100.28:3857:

SUBSCRIBE sip:192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-70m67596zq9g;rport

From: <sip:40@pbx.company.com>;tag=fglsf110n4

To: <sip:40@pbx.company.com;user=phone>;tag=b55eb6a82a

Call-ID: 253d3a3c7e37-ercmwvjp4bni

CSeq: 14830 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom870/8.4.18

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

[8] 2012/04/06 11:44:36: Packet authenticated by transport layer

[5] 2012/04/06 11:44:36: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-70m67596zq9g;rport=3857

From: <sip:40@pbx.company.com>;tag=fglsf110n4

To: <sip:40@pbx.company.com;user=phone>;tag=b55eb6a82a

Call-ID: 253d3a3c7e37-ercmwvjp4bni

CSeq: 14830 SUBSCRIBE

Contact: <sip:192.168.100.9:5061;transport=tls>

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Expires: 179

Content-Length: 0

 

[5] 2012/04/06 11:44:45: SIP Rx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-d3c7b758d1f7eb84428823e1385cecce;rport=5061

From: "unknown-caller-name" <sip:unknown-ani@pbx.company.com:5066;user=phone>;tag=477149378

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=ecutacjb4m

Call-ID: 68e01a15@pbx

CSeq: 14901 INVITE

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

User-Agent: snom870/8.4.18

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

Content-Type: application/sdp

Content-Length: 443

 

v=0

o=root 841666995 841666996 IN IP4 192.168.100.28

s=call

c=IN IP4 192.168.100.28

t=0 0

m=audio 54512 RTP/AVP 8 0 9 2 3 101

a=rtpmap:8 pcma/8000

a=rtpmap:0 pcmu/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=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Kq0LS49rw+ItTl2UM6L2Oz5HpRP7EJ9NeZ7RbN1l

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

a=sendrecv

[7] 2012/04/06 11:44:45: Call 68e01a15@pbx: Clear last INVITE

[6] 2012/04/06 11:44:45: Call-leg 2: Codec pcma/8000 is chosen for call id 68e01a15@pbx

[6] 2012/04/06 11:44:45: Call-leg 2: Sending RTP for 68e01a15@pbx to 192.168.100.28:54512, codec pcma/8000

[5] 2012/04/06 11:44:45: set codec: codec pcma/8000 is set to call-leg 2

[5] 2012/04/06 11:44:45: SIP Tx tls:192.168.100.28:3857:

ACK sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6 SIP/2.0

Via: SIP/2.0/TLS 192.168.100.9:5061;branch=z9hG4bK-359192af4fbae6cfff1afb0d98aeed1e;rport

From: "unknown-caller-name" <sip:unknown-ani@pbx.company.com:5066;user=phone>;tag=477149378

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=ecutacjb4m

Call-ID: 68e01a15@pbx

CSeq: 14901 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.100.9:5061;transport=tls>

Content-Length: 0

 

[7] 2012/04/06 11:44:45: Determine pass-through mode after receiving response

[8] 2012/04/06 11:44:45: Call state for call object 1: connected

[7] 2012/04/06 11:44:45: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil: RTP pass-through mode

[7] 2012/04/06 11:44:45: 68e01a15@pbx: RTP pass-through mode

[5] 2012/04/06 11:44:51: SIP Rx tls:192.168.100.28:3857:

BYE sip:40@192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-rgnxs54lc3d0;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=ecutacjb4m

To: "unknown-caller-name" <sip:unknown-ani@pbx.company.com:5066;user=phone>;tag=477149378

Call-ID: 68e01a15@pbx

CSeq: 1 BYE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

User-Agent: snom870/8.4.18

RTP-RxStat: Total_Rx_Pkts=272,Rx_Pkts=271,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=11659834

RTP-TxStat: Total_Tx_Pkts=272,Tx_Pkts=272,Remote_Tx_Pkts=-124725708

Proxy-Require: buttons

Content-Length: 0

 

[8] 2012/04/06 11:44:51: Packet authenticated by transport layer

[5] 2012/04/06 11:44:51: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-rgnxs54lc3d0;rport=3857

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=ecutacjb4m

To: "unknown-caller-name" <sip:unknown-ani@pbx.company.com:5066;user=phone>;tag=477149378

Call-ID: 68e01a15@pbx

CSeq: 1 BYE

Contact: <sip:40@192.168.100.9:5061;transport=tls>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[7] 2012/04/06 11:44:51: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil: Media-aware pass-through mode

[8] 2012/04/06 11:44:51: Clearing call port 2, SIP call id 68e01a15@pbx

[9] 2012/04/06 11:44:51: Resolve 19792: url sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp

[9] 2012/04/06 11:44:51: Resolve 19792: a udp 127.0.0.1 5066

[9] 2012/04/06 11:44:51: Resolve 19792: udp 127.0.0.1 5066

[5] 2012/04/06 11:44:51: SIP Tx udp:127.0.0.1:5066:

BYE sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-7a593e08d323e101558270a1338bddc4;rport

From: "342410" <sip:342410@localhost:5060>;tag=d8365b0f5d

To: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-634854-1841585795-105ds-1c69fb81-faaca7f0

Call-ID: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

CSeq: 28781 BYE

Max-Forwards: 70

Contact: <sip:342410@127.0.0.1:5060;transport=udp>

Content-Length: 0

 

[5] 2012/04/06 11:44:51: SIP Rx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-7a593e08d323e101558270a1338bddc4;rport=5060

From: "342410" <sip:342410@localhost:5060>;tag=d8365b0f5d

To: "unknown-caller-name" <sip:unknown-ani@192.168.100.9:5066>;tag=pxip-callid-1333705434-634854-1841585795-105ds-1c69fb81-faaca7f0

Call-ID: 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

CSeq: 28781 BYE

Content-Length: 0

 

[7] 2012/04/06 11:44:51: Call 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil: Clear last request

[5] 2012/04/06 11:44:51: BYE Response: Terminate 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

[8] 2012/04/06 11:44:51: Clearing call port 0, SIP call id 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

[8] 2012/04/06 11:44:51: Remove leg 3: call port 2, SIP call id 68e01a15@pbx

[8] 2012/04/06 11:44:51: Hangup: Call 2 not found

[8] 2012/04/06 11:44:51: Last message repeated 2 times

[8] 2012/04/06 11:44:51: Remove leg 1: call port 0, SIP call id 063116d2-1dd2-11b2-8d9f-b74d1ca2386a@snom.home.nil

[5] 2012/04/06 11:44:57: SIP Rx tls:192.168.100.28:3857:

SUBSCRIBE sip:192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-4za6b3j2nq8u;rport

From: <sip:40@pbx.company.com>;tag=xh6zdvdigt

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=0ceb9e1a8b

Call-ID: 153d3a3cc317-b7c03ro5zct8

CSeq: 14820 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

Event: as-feature-event

User-Agent: snom870/8.4.18

Proxy-Require: buttons

Expires: 3600

Content-Type: application/x-as-feature-event+xml

Content-Length: 0

 

[8] 2012/04/06 11:44:57: Packet authenticated by transport layer

[5] 2012/04/06 11:44:57: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-4za6b3j2nq8u;rport=3857

From: <sip:40@pbx.company.com>;tag=xh6zdvdigt

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=0ceb9e1a8b

Call-ID: 153d3a3cc317-b7c03ro5zct8

CSeq: 14820 SUBSCRIBE

Contact: <sip:192.168.100.9:5061;transport=tls>

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Expires: 179

Content-Length: 0

 

[5] 2012/04/06 11:45:33: SIP Rx tls:192.168.100.28:3857:

REGISTER sip:pbx.company.com SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-0684s4sp4woy;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=jp1f4inbim

To: "Walter Hackner" <sip:40@pbx.company.com>

Call-ID: 3e70263c147b-u4cbzoqqc1vt

CSeq: 32601 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:2dfbe0a8-1848-4d08-9261-4b8e55184131>"

User-Agent: snom870/8.4.18

Allow-Events: dialog

X-Real-IP: 192.168.100.28

Supported: path, gruu

WWW-Contact: <http://192.168.100.28:80>

WWW-Contact: <https://192.168.100.28:443>

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

[8] 2012/04/06 11:45:33: Packet authenticated by transport layer

[5] 2012/04/06 11:45:33: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-0684s4sp4woy;rport=3857

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=jp1f4inbim

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=f11c9ea38f

Call-ID: 3e70263c147b-u4cbzoqqc1vt

CSeq: 32601 REGISTER

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;expires=182

Supported: path

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[5] 2012/04/06 11:45:54: SIP Rx udp:127.0.0.1:5066:

OPTIONS sip:localhost:5060;transport=udp SIP/2.0

From: <sip:192.168.100.9:5066>;tag=ds-37cc71e1-174a399c

To: <sip:localhost:5060>

Contact: <sip:127.0.0.1:5066>;transport=udp

Call-ID: 4dbad9a2-1dd2-11b2-b219-cac62cfdeb6e@snom.home.nil

CSeq: 12771475 OPTIONS

Content-Length: 0

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bK4dbb9112-1dd2-11b2-a46a-a88f8c30eb2e

Max-Forwards: 70

 

[9] 2012/04/06 11:45:54: Resolve 19795: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:45:54: Resolve 19795: a udp 127.0.0.1 5066

[9] 2012/04/06 11:45:54: Resolve 19795: udp 127.0.0.1 5066

[5] 2012/04/06 11:45:54: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK4dbb9112-1dd2-11b2-a46a-a88f8c30eb2e

From: <sip:192.168.100.9:5066>;tag=ds-37cc71e1-174a399c

To: <sip:localhost:5060>;tag=f25a3c0857

Call-ID: 4dbad9a2-1dd2-11b2-b219-cac62cfdeb6e@snom.home.nil

CSeq: 12771475 OPTIONS

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

Content-Length: 0

 

[5] 2012/04/06 11:46:05: SIP Rx tls:192.168.100.28:3857:

SUBSCRIBE sip:192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-qc3mfhofogk0;rport

From: <sip:40@pbx.company.com>;tag=fglsf110n4

To: <sip:40@pbx.company.com;user=phone>;tag=b55eb6a82a

Call-ID: 253d3a3c7e37-ercmwvjp4bni

CSeq: 14831 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom870/8.4.18

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

[8] 2012/04/06 11:46:05: Packet authenticated by transport layer

[5] 2012/04/06 11:46:05: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-qc3mfhofogk0;rport=3857

From: <sip:40@pbx.company.com>;tag=fglsf110n4

To: <sip:40@pbx.company.com;user=phone>;tag=b55eb6a82a

Call-ID: 253d3a3c7e37-ercmwvjp4bni

CSeq: 14831 SUBSCRIBE

Contact: <sip:192.168.100.9:5061;transport=tls>

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Expires: 179

Content-Length: 0

 

[5] 2012/04/06 11:46:26: SIP Rx tls:192.168.100.28:3857:

SUBSCRIBE sip:192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-lwp9cy8z2up8;rport

From: <sip:40@pbx.company.com>;tag=xh6zdvdigt

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=0ceb9e1a8b

Call-ID: 153d3a3cc317-b7c03ro5zct8

CSeq: 14821 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

Event: as-feature-event

User-Agent: snom870/8.4.18

Proxy-Require: buttons

Expires: 3600

Content-Type: application/x-as-feature-event+xml

Content-Length: 0

 

[8] 2012/04/06 11:46:26: Packet authenticated by transport layer

[5] 2012/04/06 11:46:26: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-lwp9cy8z2up8;rport=3857

From: <sip:40@pbx.company.com>;tag=xh6zdvdigt

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=0ceb9e1a8b

Call-ID: 153d3a3cc317-b7c03ro5zct8

CSeq: 14821 SUBSCRIBE

Contact: <sip:192.168.100.9:5061;transport=tls>

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Expires: 181

Content-Length: 0

 

[5] 2012/04/06 11:47:04: SIP Rx tls:192.168.100.28:3857:

REGISTER sip:pbx.company.com SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-x4pl3x27x736;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=nhfjs80jkv

To: "Walter Hackner" <sip:40@pbx.company.com>

Call-ID: 3e70263c147b-u4cbzoqqc1vt

CSeq: 32603 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:2dfbe0a8-1848-4d08-9261-4b8e55184131>"

User-Agent: snom870/8.4.18

Allow-Events: dialog

X-Real-IP: 192.168.100.28

Supported: path, gruu

WWW-Contact: <http://192.168.100.28:80>

WWW-Contact: <https://192.168.100.28:443>

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

[8] 2012/04/06 11:47:04: Packet authenticated by transport layer

[5] 2012/04/06 11:47:04: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-x4pl3x27x736;rport=3857

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=nhfjs80jkv

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=f11c9ea38f

Call-ID: 3e70263c147b-u4cbzoqqc1vt

CSeq: 32603 REGISTER

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;expires=181

Supported: path

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[6] 2012/04/06 11:47:27: Received bindRequest for user pbx.company.com\40

[5] 2012/04/06 11:47:34: SIP Rx tls:192.168.100.28:3857:

SUBSCRIBE sip:192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-n0db9c2v0l5d;rport

From: <sip:40@pbx.company.com>;tag=fglsf110n4

To: <sip:40@pbx.company.com;user=phone>;tag=b55eb6a82a

Call-ID: 253d3a3c7e37-ercmwvjp4bni

CSeq: 14832 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom870/8.4.18

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

[8] 2012/04/06 11:47:34: Packet authenticated by transport layer

[5] 2012/04/06 11:47:34: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-n0db9c2v0l5d;rport=3857

From: <sip:40@pbx.company.com>;tag=fglsf110n4

To: <sip:40@pbx.company.com;user=phone>;tag=b55eb6a82a

Call-ID: 253d3a3c7e37-ercmwvjp4bni

CSeq: 14832 SUBSCRIBE

Contact: <sip:192.168.100.9:5061;transport=tls>

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Expires: 179

Content-Length: 0

 

[6] 2012/04/06 11:47:35: Received searchRequest(type 128), substrings=+1796946824

[5] 2012/04/06 11:47:35: SIP Rx tls:192.168.100.28:3857:

INVITE sip:001796946824@pbx.company.com;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-xvhbzmk4ragh;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=6m2w0b8pyt

To: <sip:001796946824@pbx.company.com;user=phone>

Call-ID: b4db3d3c3c3e-dlzm72jtag0x

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

X-Serialnumber: 000413412514

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

User-Agent: snom870/8.4.18

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: 526

 

v=0

o=root 871259571 871259571 IN IP4 192.168.100.28

s=call

c=IN IP4 192.168.100.28

t=0 0

m=audio 65420 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:c3+vzhHDgQ5VHF0sOhTlfPnFegQtA2euNP326DrH

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] 2012/04/06 11:47:35: Packet authenticated by transport layer

[8] 2012/04/06 11:47:35: Allocating for call port 3, SIP call id b4db3d3c3c3e-dlzm72jtag0x

[9] 2012/04/06 11:47:35: UDP(IPv4): Opening socket on 0.0.0.0:51758

[9] 2012/04/06 11:47:35: UDP(IPv4): Opening socket on 0.0.0.0:51759

[9] 2012/04/06 11:47:35: UDP(IPv6): Opening socket on [::]:51758

[9] 2012/04/06 11:47:35: UDP(IPv6): Opening socket on [::]:51759

[8] 2012/04/06 11:47:35: Could not find a trunk (3 trunks)

[9] 2012/04/06 11:47:35: Using outbound proxy sip:192.168.100.28:3857;transport=tls because of flow-label

[9] 2012/04/06 11:47:35: Last message repeated 3 times

[5] 2012/04/06 11:47:35: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-xvhbzmk4ragh;rport=3857

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=6m2w0b8pyt

To: <sip:001796946824@pbx.company.com;user=phone>;tag=31a39c80d1

Call-ID: b4db3d3c3c3e-dlzm72jtag0x

CSeq: 1 INVITE

Content-Length: 0

 

[7] 2012/04/06 11:47:35: Set packet length to 20

[6] 2012/04/06 11:47:35: Call-leg 3: Sending RTP for b4db3d3c3c3e-dlzm72jtag0x to 192.168.100.28:65420, codec not set yet

[8] 2012/04/06 11:47:35: Incoming call: Request URI sip:001796946824@pbx.company.com;user=phone, To is <sip:001796946824@pbx.company.com;user=phone>

[8] 2012/04/06 11:47:35: Call from an user 40

[8] 2012/04/06 11:47:35: To is <sip:001796946824@pbx.company.com;user=phone>, user 0, domain 1

[8] 2012/04/06 11:47:35: From user 40

[8] 2012/04/06 11:47:35: Set the To domain based on From user 40@pbx.company.com

[8] 2012/04/06 11:47:35: Call state for call object 3: idle

[7] 2012/04/06 11:47:35: Call port 3: set_codecs for b4db3d3c3c3e-dlzm72jtag0x codecs "", codec_preference count 6

[9] 2012/04/06 11:47:35: Dialplan: Evaluating !^99([0-9]*)@.*!sip:\1@\r;user=phone!i against 001796946824@pbx.company.com

[6] 2012/04/06 11:47:35: The trunk sipgate is disabled. Skipping it...

[9] 2012/04/06 11:47:35: Dialplan: Evaluating !^0([0-9]*)@.*!sip:\1@\r;user=phone!i against 001796946824@pbx.company.com

[5] 2012/04/06 11:47:35: Dialplan "Hackner": Match 001796946824@pbx.company.com to sip:01796946824@127.0.0.1:5066;user=phone on trunk NBE

[9] 2012/04/06 11:47:35: Generating hf header using {from}

[9] 2012/04/06 11:47:35: Generating ht header using {to}

[9] 2012/04/06 11:47:35: Generating hppi header using {trunk}

[8] 2012/04/06 11:47:35: Play audio_moh/noise.wav, caching true

[8] 2012/04/06 11:47:35: Allocating for call port 4, SIP call id a200ab31@pbx

[9] 2012/04/06 11:47:35: UDP(IPv4): Opening socket on 0.0.0.0:58272

[9] 2012/04/06 11:47:35: UDP(IPv4): Opening socket on 0.0.0.0:58273

[9] 2012/04/06 11:47:35: UDP(IPv6): Opening socket on [::]:58272

[9] 2012/04/06 11:47:35: UDP(IPv6): Opening socket on [::]:58273

[7] 2012/04/06 11:47:35: Call port 4: set_codecs for a200ab31@pbx codecs "8", codec_preference count 2

[9] 2012/04/06 11:47:35: Call port 4: update_codecs for a200ab31@pbx: adding codec pcma/8000 to available list

[9] 2012/04/06 11:47:35: Call port 4: update_codecs for a200ab31@pbx: codec_preference size 2, available codecs size 2

[9] 2012/04/06 11:47:35: Resolve 19801: url sip:127.0.0.1:5066

[9] 2012/04/06 11:47:35: Resolve 19801: udp 127.0.0.1 5066

[5] 2012/04/06 11:47:35: SIP Tx udp:127.0.0.1:5066:

INVITE sip:01796946824@127.0.0.1:5066;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-ebee290edecac17c35b631e2f32299df;rport

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=313954159

To: <sip:001796946824@pbx.company.com;user=phone>

Call-ID: a200ab31@pbx

CSeq: 23389 INVITE

Max-Forwards: 70

Contact: <sip:40@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

P-Preferred-Identity: "pbx.company.com" <sip:127.0.0.1:5066>

P-Charging-Vector: icid-value=;icid-generated-at=127.0.0.1;orig-ioi=pbx.company.com

Content-Type: application/sdp

Content-Length: 233

 

v=0

o=- 1081022980 1081022980 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 58272 RTP/AVP 8 101

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

[7] 2012/04/06 11:47:35: Set packet length to 20

[9] 2012/04/06 11:47:35: Call port 3: update_codecs for b4db3d3c3c3e-dlzm72jtag0x: adding codec pcma/8000 to available list

[9] 2012/04/06 11:47:35: Call port 3: update_codecs for b4db3d3c3c3e-dlzm72jtag0x: adding codec pcmu/8000 to available list

[9] 2012/04/06 11:47:35: Call port 3: update_codecs for b4db3d3c3c3e-dlzm72jtag0x: adding codec g722/8000 to available list

[9] 2012/04/06 11:47:35: Call port 3: update_codecs for b4db3d3c3c3e-dlzm72jtag0x: adding codec g726-32/8000 to available list

[9] 2012/04/06 11:47:35: Call port 3: update_codecs for b4db3d3c3c3e-dlzm72jtag0x: adding codec gsm/8000 to available list

[9] 2012/04/06 11:47:35: Call port 3: update_codecs for b4db3d3c3c3e-dlzm72jtag0x: codec_preference size 6, available codecs size 6

[6] 2012/04/06 11:47:35: Call-leg 3: Codec pcma/8000 is chosen for call id b4db3d3c3c3e-dlzm72jtag0x

[5] 2012/04/06 11:47:35: set codec: codec pcma/8000 is set to call-leg 3

[5] 2012/04/06 11:47:35: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-xvhbzmk4ragh;rport=3857

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=6m2w0b8pyt

To: <sip:001796946824@pbx.company.com;user=phone>;tag=31a39c80d1

Call-ID: b4db3d3c3c3e-dlzm72jtag0x

CSeq: 1 INVITE

Contact: <sip:40@192.168.100.9:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 433

 

v=0

o=- 781422400 781422400 IN IP4 192.168.100.9

s=-

c=IN IP4 192.168.100.9

t=0 0

m=audio 51758 RTP/AVP 8 0 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8JbnXE7PVrhLg6WUeg8VQ23dZLkamdmUag3amL0t

a=rtpmap:8 pcma/8000

a=rtpmap:0 pcmu/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

[5] 2012/04/06 11:47:35: SIP Rx udp:127.0.0.1:5066:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-ebee290edecac17c35b631e2f32299df;rport=5060

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=313954159

To: <sip:001796946824@pbx.company.com;user=phone>;tag=ds-487806bf-2f64ccd0

Call-ID: a200ab31@pbx

CSeq: 23389 INVITE

Content-Length: 0

Server: Netborder Express Gateway/4.1.6

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

 

[9] 2012/04/06 11:47:35: Message repetition, packet dropped

[6] 2012/04/06 11:47:35: Received searchRequest(type 128), substrings=001796946824

[5] 2012/04/06 11:47:36: SIP Rx tls:192.168.100.28:3857:

PRACK sip:40@192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-d4zdttyoxcyr;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=6m2w0b8pyt

To: <sip:001796946824@pbx.company.com;user=phone>;tag=31a39c80d1

Call-ID: b4db3d3c3c3e-dlzm72jtag0x

CSeq: 2 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;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] 2012/04/06 11:47:36: Packet authenticated by transport layer

[5] 2012/04/06 11:47:36: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-d4zdttyoxcyr;rport=3857

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=6m2w0b8pyt

To: <sip:001796946824@pbx.company.com;user=phone>;tag=31a39c80d1

Call-ID: b4db3d3c3c3e-dlzm72jtag0x

CSeq: 2 PRACK

Contact: <sip:40@192.168.100.9:5061;transport=tls>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[5] 2012/04/06 11:47:41: SIP Rx udp:127.0.0.1:5066:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-ebee290edecac17c35b631e2f32299df;rport=5060

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=313954159

To: <sip:001796946824@pbx.company.com;user=phone>;tag=ds-487806bf-2f64ccd0

Call-ID: a200ab31@pbx

CSeq: 23389 INVITE

Content-Length: 0

Server: Netborder Express Gateway/4.1.6

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

 

[8] 2012/04/06 11:47:41: Call state for call object 3: alerting

[6] 2012/04/06 11:47:41: Trunk NBE: Ignoring the SDP due to the trunk setting

[8] 2012/04/06 11:47:41: Play audio_de/ringback.wav, caching true

[5] 2012/04/06 11:47:45: SIP Rx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-ebee290edecac17c35b631e2f32299df;rport=5060

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=313954159

To: <sip:001796946824@pbx.company.com;user=phone>;tag=ds-487806bf-2f64ccd0

Call-ID: a200ab31@pbx

CSeq: 23389 INVITE

Content-Length: 230

Content-Type: application/sdp

Contact: <sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp>

Server: Netborder Express Gateway/4.1.6

 

v=0

o=Sangoma-Tech 1333705665 1333705714 IN IP4 127.0.0.1

s=SIP Call

c=IN IP4 192.168.100.9

t=0 0

m=audio 14004 RTP/AVP 8 101

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

[7] 2012/04/06 11:47:45: Call a200ab31@pbx: Clear last INVITE

[7] 2012/04/06 11:47:45: Set packet length to 20

[6] 2012/04/06 11:47:45: Call-leg 4: Codec pcma/8000 is chosen for call id a200ab31@pbx

[6] 2012/04/06 11:47:45: Call-leg 4: Sending RTP for a200ab31@pbx to 192.168.100.9:14004, codec pcma/8000

[5] 2012/04/06 11:47:45: set codec: codec pcma/8000 is set to call-leg 4

[9] 2012/04/06 11:47:45: Resolve 19804: url sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp

[9] 2012/04/06 11:47:45: Resolve 19804: a udp 127.0.0.1 5066

[9] 2012/04/06 11:47:45: Resolve 19804: udp 127.0.0.1 5066

[5] 2012/04/06 11:47:45: SIP Tx udp:127.0.0.1:5066:

ACK sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-2a23cb037b6ac0effc705b4e6f537e55;rport

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=313954159

To: <sip:001796946824@pbx.company.com;user=phone>;tag=ds-487806bf-2f64ccd0

Call-ID: a200ab31@pbx

CSeq: 23389 ACK

Max-Forwards: 70

Contact: <sip:40@127.0.0.1:5060;transport=udp>

P-Preferred-Identity: "pbx.company.com" <sip:127.0.0.1:5066>

P-Charging-Vector: icid-value=;icid-generated-at=127.0.0.1;orig-ioi=pbx.company.com

Content-Length: 0

 

[7] 2012/04/06 11:47:45: Determine pass-through mode after receiving response

[8] 2012/04/06 11:47:45: Call state for call object 3: connected

[5] 2012/04/06 11:47:45: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-xvhbzmk4ragh;rport=3857

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=6m2w0b8pyt

To: <sip:001796946824@pbx.company.com;user=phone>;tag=31a39c80d1

Call-ID: b4db3d3c3c3e-dlzm72jtag0x

CSeq: 1 INVITE

Contact: <sip:40@192.168.100.9:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Type: application/sdp

Content-Length: 433

 

v=0

o=- 781422400 781422400 IN IP4 192.168.100.9

s=-

c=IN IP4 192.168.100.9

t=0 0

m=audio 51758 RTP/AVP 8 0 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8JbnXE7PVrhLg6WUeg8VQ23dZLkamdmUag3amL0t

a=rtpmap:8 pcma/8000

a=rtpmap:0 pcmu/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] 2012/04/06 11:47:45: b4db3d3c3c3e-dlzm72jtag0x: RTP pass-through mode

[7] 2012/04/06 11:47:45: a200ab31@pbx: RTP pass-through mode

[6] 2012/04/06 11:47:45: Call-leg 4: Sending RTP for a200ab31@pbx to 127.0.0.1:14004, codec pcma/8000

[5] 2012/04/06 11:47:45: SIP Rx tls:192.168.100.28:3857:

ACK sip:40@192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-l67pbhf8wg3v;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=6m2w0b8pyt

To: <sip:001796946824@pbx.company.com;user=phone>;tag=31a39c80d1

Call-ID: b4db3d3c3c3e-dlzm72jtag0x

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

 

[8] 2012/04/06 11:47:45: Packet authenticated by transport layer

[5] 2012/04/06 11:47:54: SIP Rx udp:127.0.0.1:5066:

OPTIONS sip:localhost:5060;transport=udp SIP/2.0

From: <sip:192.168.100.9:5066>;tag=ds-231f7956-33e6e3b0

To: <sip:localhost:5060>

Contact: <sip:127.0.0.1:5066>;transport=udp

Call-ID: 95427eb0-1dd1-11b2-b82c-9da84ca40c4b@snom.home.nil

CSeq: 11973243 OPTIONS

Content-Length: 0

Via: SIP/2.0/UDP 127.0.0.1:5066;rport;branch=z9hG4bK9542bdd0-1dd1-11b2-b5b3-c0e112607e09

Max-Forwards: 70

 

[9] 2012/04/06 11:47:54: Resolve 19806: aaaa udp 127.0.0.1 5066

[9] 2012/04/06 11:47:54: Resolve 19806: a udp 127.0.0.1 5066

[9] 2012/04/06 11:47:54: Resolve 19806: udp 127.0.0.1 5066

[5] 2012/04/06 11:47:54: SIP Tx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bK9542bdd0-1dd1-11b2-b5b3-c0e112607e09

From: <sip:192.168.100.9:5066>;tag=ds-231f7956-33e6e3b0

To: <sip:localhost:5060>;tag=8684adce74

Call-ID: 95427eb0-1dd1-11b2-b82c-9da84ca40c4b@snom.home.nil

CSeq: 11973243 OPTIONS

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

Content-Length: 0

 

[5] 2012/04/06 11:47:56: SIP Rx tls:192.168.100.28:3857:

SUBSCRIBE sip:192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-37keh0ff4fsi;rport

From: <sip:40@pbx.company.com>;tag=xh6zdvdigt

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=0ceb9e1a8b

Call-ID: 153d3a3cc317-b7c03ro5zct8

CSeq: 14822 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

Event: as-feature-event

User-Agent: snom870/8.4.18

Proxy-Require: buttons

Expires: 3600

Content-Type: application/x-as-feature-event+xml

Content-Length: 0

 

[8] 2012/04/06 11:47:56: Packet authenticated by transport layer

[5] 2012/04/06 11:47:56: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-37keh0ff4fsi;rport=3857

From: <sip:40@pbx.company.com>;tag=xh6zdvdigt

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=0ceb9e1a8b

Call-ID: 153d3a3cc317-b7c03ro5zct8

CSeq: 14822 SUBSCRIBE

Contact: <sip:192.168.100.9:5061;transport=tls>

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Expires: 179

Content-Length: 0

 

[5] 2012/04/06 11:48:03: SIP Rx tls:192.168.100.28:3857:

BYE sip:40@192.168.100.9:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-saqpmo1re81p;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=6m2w0b8pyt

To: <sip:001796946824@pbx.company.com;user=phone>;tag=31a39c80d1

Call-ID: b4db3d3c3c3e-dlzm72jtag0x

CSeq: 3 BYE

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1

User-Agent: snom870/8.4.18

RTP-RxStat: Total_Rx_Pkts=1383,Rx_Pkts=1363,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=2279428

RTP-TxStat: Total_Tx_Pkts=1365,Tx_Pkts=1365,Remote_Tx_Pkts=-39100101

Proxy-Require: buttons

Content-Length: 0

 

[8] 2012/04/06 11:48:03: Packet authenticated by transport layer

[5] 2012/04/06 11:48:03: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-saqpmo1re81p;rport=3857

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=6m2w0b8pyt

To: <sip:001796946824@pbx.company.com;user=phone>;tag=31a39c80d1

Call-ID: b4db3d3c3c3e-dlzm72jtag0x

CSeq: 3 BYE

Contact: <sip:40@192.168.100.9:5061;transport=tls>

User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

[7] 2012/04/06 11:48:03: a200ab31@pbx: Media-aware pass-through mode

[8] 2012/04/06 11:48:03: Clearing call port 3, SIP call id b4db3d3c3c3e-dlzm72jtag0x

[9] 2012/04/06 11:48:03: Resolve 19809: url sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp

[9] 2012/04/06 11:48:03: Resolve 19809: a udp 127.0.0.1 5066

[9] 2012/04/06 11:48:03: Resolve 19809: udp 127.0.0.1 5066

[8] 2012/04/06 11:48:03: Remove leg 4: call port 3, SIP call id b4db3d3c3c3e-dlzm72jtag0x

[5] 2012/04/06 11:48:03: SIP Tx udp:127.0.0.1:5066:

BYE sip:NetborderExpressGateway@127.0.0.1:5066;transport=udp SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-9fc9c53411c2b37897b3bffcfc04c595;rport

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=313954159

To: <sip:001796946824@pbx.company.com;user=phone>;tag=ds-487806bf-2f64ccd0

Call-ID: a200ab31@pbx

CSeq: 23390 BYE

Max-Forwards: 70

Contact: <sip:40@127.0.0.1:5060;transport=udp>

P-Preferred-Identity: "pbx.company.com" <sip:127.0.0.1:5066>

P-Charging-Vector: icid-value=;icid-generated-at=127.0.0.1;orig-ioi=pbx.company.com

Content-Length: 0

 

[8] 2012/04/06 11:48:03: Hangup: Call 3 not found

[8] 2012/04/06 11:48:03: Last message repeated 2 times

[5] 2012/04/06 11:48:03: SIP Rx udp:127.0.0.1:5066:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-9fc9c53411c2b37897b3bffcfc04c595;rport=5060

From: "Walter Hackner" <sip:40@pbx.company.com;user=phone>;tag=313954159

To: <sip:001796946824@pbx.company.com;user=phone>;tag=ds-487806bf-2f64ccd0

Call-ID: a200ab31@pbx

CSeq: 23390 BYE

Content-Length: 0

 

[7] 2012/04/06 11:48:03: Call a200ab31@pbx: Clear last request

[5] 2012/04/06 11:48:03: BYE Response: Terminate a200ab31@pbx

[8] 2012/04/06 11:48:03: Clearing call port 4, SIP call id a200ab31@pbx

[8] 2012/04/06 11:48:03: Remove leg 5: call port 4, SIP call id a200ab31@pbx

[9] 2012/04/06 11:48:04: Remote site 192.168.100.28 closed the connection

[5] 2012/04/06 11:48:34: SIP Rx tls:192.168.100.28:3857:

REGISTER sip:pbx.company.com SIP/2.0

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-it5h8tglb9v1;rport

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=p4ttxggzg4

To: "Walter Hackner" <sip:40@pbx.company.com>

Call-ID: 3e70263c147b-u4cbzoqqc1vt

CSeq: 32605 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:2dfbe0a8-1848-4d08-9261-4b8e55184131>"

User-Agent: snom870/8.4.18

Allow-Events: dialog

X-Real-IP: 192.168.100.28

Supported: path, gruu

WWW-Contact: <http://192.168.100.28:80>

WWW-Contact: <https://192.168.100.28:443>

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

[8] 2012/04/06 11:48:34: Packet authenticated by transport layer

[5] 2012/04/06 11:48:34: SIP Tx tls:192.168.100.28:3857:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.28:3857;branch=z9hG4bK-it5h8tglb9v1;rport=3857

From: "Walter Hackner" <sip:40@pbx.company.com>;tag=p4ttxggzg4

To: "Walter Hackner" <sip:40@pbx.company.com>;tag=f11c9ea38f

Call-ID: 3e70263c147b-u4cbzoqqc1vt

CSeq: 32605 REGISTER

Contact: <sip:40@192.168.100.28:3857;transport=tls;line=0ii3wcy6>;expires=182

Supported: path

Server: snomONE/2011-4.5.0.1030 Beta Corona Austrinids

Content-Length: 0

 

 

Das Ergbenis ist wieder gleich: unverständliche Sprachqualität!

Zusätzlich habe ich am NBE das Call Recording aktiviert. Auf den dort aufgezeichneten Sounddateien ist die Sprachqualität jedoch glasklar!

Ich bitte um weitere Hilfe!

1333705434-627575-561717988-103_pstn_unknown_tx.wav

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