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When placing an outbound call using a cell phone, it is still showing the cell phone number as the caller ID, I thought this was supposed to be fixed in this version. 3.1.1.3113 (Win32), I tied with 2 carriers, Broadvox and Vitelity, all other caller ID functions are working correctly, does it have to do with Remote Party/Privacy Indication:? I tried several setting they all show the cell phone number as the caller ID, except no indication, which shows unavailabe.

 

[7] 2008/12/25 11:23:17: SIP Rx udp:64.2.142.30:5060:

INVITE sip:2124007000@192.168.10.2 SIP/2.0

Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK02616f7b;rport

From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c

To: <sip:2124007000@216.112.126.83>

Contact: <sip:9175551212@64.2.142.30>

Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Thu, 25 Dec 2008 16:23:23 GMT

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

Content-Type: application/sdp

Content-Length: 306

 

v=0

o=root 4115 4115 IN IP4 64.2.142.30

s=session

c=IN IP4 64.2.142.30

t=0 0

m=audio 11768 RTP/AVP 0 8 3 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

 

[9] 2008/12/25 11:23:17: UDP: Opening socket on port 58244

[9] 2008/12/25 11:23:17: UDP: Opening socket on port 58245

[5] 2008/12/25 11:23:17: Identify trunk (IP address and DID match) 5

[9] 2008/12/25 11:23:17: Resolve 50145: aaaa udp 64.2.142.30 5060

[9] 2008/12/25 11:23:17: Resolve 50145: a udp 64.2.142.30 5060

[9] 2008/12/25 11:23:17: Resolve 50145: udp 64.2.142.30 5060

[7] 2008/12/25 11:23:17: SIP Tx udp:64.2.142.30:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK02616f7b;rport=5060

From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c

To: <sip:2124007000@216.112.126.83>;tag=09c02f6724

Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30

CSeq: 102 INVITE

Content-Length: 0

 

 

[6] 2008/12/25 11:23:17: Sending RTP for 29f00291573171d169e7ada9613cbd6a@64.2.142.30#09c02f6724 to 64.2.142.30:11768

[5] 2008/12/25 11:23:17: Trunk vitelity inbound sends call to +12124007000 in domain localhost

[7] 2008/12/25 11:23:17: Received call from cell phone +19175551212

[8] 2008/12/25 11:23:17: Play audio_en/aa_outbound.wav audio_en/bi_press_1.wav audio_en/aa_goto_mailbox.wav audio_en/bi_press_2.wav audio_en/aa_goto_attendant.wav audio_en/bi_press_3.wav space50

[9] 2008/12/25 11:23:17: Resolve 50146: aaaa udp 64.2.142.30 5060

[9] 2008/12/25 11:23:17: Resolve 50146: a udp 64.2.142.30 5060

[9] 2008/12/25 11:23:17: Resolve 50146: udp 64.2.142.30 5060

[7] 2008/12/25 11:23:17: SIP Tx udp:64.2.142.30:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK02616f7b;rport=5060

From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c

To: <sip:2124007000@216.112.126.83>;tag=09c02f6724

Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30

CSeq: 102 INVITE

Contact: <sip:torn_pbx@192.168.10.2:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.1.1.3113

Content-Type: application/sdp

Content-Length: 286

 

v=0

o=- 22539 22539 IN IP4 192.168.10.2

s=-

c=IN IP4 192.168.10.2

t=0 0

m=audio 58244 RTP/AVP 0 8 18 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[9] 2008/12/25 11:23:17: Resolve 50147: aaaa udp 64.2.142.30 5060

[9] 2008/12/25 11:23:17: Resolve 50147: a udp 64.2.142.30 5060

[9] 2008/12/25 11:23:17: Resolve 50147: udp 64.2.142.30 5060

[7] 2008/12/25 11:23:17: SIP Tx udp:64.2.142.30:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK02616f7b;rport=5060

From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c

To: <sip:2124007000@216.112.126.83>;tag=09c02f6724

Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30

CSeq: 102 INVITE

Contact: <sip:torn_pbx@192.168.10.2:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.1.1.3113

Content-Type: application/sdp

Content-Length: 286

 

v=0

o=- 22539 22539 IN IP4 192.168.10.2

s=-

c=IN IP4 192.168.10.2

t=0 0

m=audio 58244 RTP/AVP 0 8 18 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/12/25 11:23:17: SIP Rx udp:64.2.142.30:5060:

ACK sip:torn_pbx@192.168.10.2:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK249a5232;rport

From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c

To: <sip:2124007000@216.112.126.83>;tag=09c02f6724

Contact: <sip:9175551212@64.2.142.30>

Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0

 

 

[7] 2008/12/25 11:23:17: SIP Rx udp:64.2.142.30:5060:

ACK sip:torn_pbx@192.168.10.2:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK306ceaaf;rport

From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c

To: <sip:2124007000@216.112.126.83>;tag=09c02f6724

Contact: <sip:9175551212@64.2.142.30>

Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0

 

 

[9] 2008/12/25 11:23:17: Message repetition, packet dropped

[6] 2008/12/25 11:23:20: Received DTMF 1

[8] 2008/12/25 11:23:20: Play audio_en/ex_enter_access_code.wav

[8] 2008/12/25 11:23:22: Play space20

[6] 2008/12/25 11:23:23: Received DTMF 1

[6] 2008/12/25 11:23:23: Received DTMF 9

[7] 2008/12/25 11:23:24: SIP Rx udp:66.114.64.65:2048:

SUBSCRIBE sip:401@sip.solutions.com;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-7vbhqhxh0w98;rport

From: <sip:401@sip.solutions.com>;tag=gren4n4yrj

To: <sip:401@sip.solutions.com;user=phone>;tag=aad9a04baf

Call-ID: 3c26701c35a2-j1weli7digqg

CSeq: 2533 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:401@192.168.81.102:2048;line=7uuu69wp>;reg-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom360/7.3.10a

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[9] 2008/12/25 11:23:24: Resolve 50148: udp 66.114.64.65 2048

[7] 2008/12/25 11:23:24: SIP Tx udp:66.114.64.65:2048:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-7vbhqhxh0w98;rport=2048;received=66.114.64.65

From: <sip:401@sip.solutions.com>;tag=gren4n4yrj

To: <sip:401@sip.solutions.com;user=phone>;tag=aad9a04baf

Call-ID: 3c26701c35a2-j1weli7digqg

CSeq: 2533 SUBSCRIBE

Contact: <sip:192.168.10.2:5060;transport=udp>

Expires: 32

Content-Length: 0

 

 

[7] 2008/12/25 11:23:24: SIP Rx udp:66.114.64.65:2048:

REGISTER sip:sip.solutions.com SIP/2.0

Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-txke68p6cw17;rport

From: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=fkb206hmbx

To: "Moishe Grunstein" <sip:401@sip.solutions.com>

Call-ID: 3c26701b6b59-zzy3nchymw3e

CSeq: 5022 REGISTER

Max-Forwards: 70

Contact: <sip:401@192.168.81.102:2048;line=7uuu69wp>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:7035dcbf-7d47-44e4-9f8a-06b3fb57c1b6>"

Contact: <http://192.168.81.102:80>

Contact: <https://192.168.81.102:443>

User-Agent: snom360/7.3.10a

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.81.102

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[9] 2008/12/25 11:23:24: Resolve 50149: udp 66.114.64.65 2048

[7] 2008/12/25 11:23:24: SIP Tx udp:66.114.64.65:2048:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-txke68p6cw17;rport=2048;received=66.114.64.65

From: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=fkb206hmbx

To: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=673192d660

Call-ID: 3c26701b6b59-zzy3nchymw3e

CSeq: 5022 REGISTER

Contact: <sip:401@192.168.81.102:2048;line=7uuu69wp>;expires=28

Contact: <http://192.168.81.102:80>;expires=28

Contact: <https://192.168.81.102:443>;expires=28

Content-Length: 0

 

 

[6] 2008/12/25 11:23:24: Received DTMF 7

[6] 2008/12/25 11:23:25: Received DTMF 6

[8] 2008/12/25 11:23:25: Play audio_en/ex_enter_number.wav

[6] 2008/12/25 11:23:27: Received DTMF 7

[6] 2008/12/25 11:23:28: Received DTMF 1

[6] 2008/12/25 11:23:29: Received DTMF 8

[6] 2008/12/25 11:23:29: Received DTMF 4

[6] 2008/12/25 11:23:30: Received DTMF 3

[6] 2008/12/25 11:23:31: Received DTMF 6

[6] 2008/12/25 11:23:32: Received DTMF 5

[6] 2008/12/25 11:23:32: Received DTMF 5

[6] 2008/12/25 11:23:34: Received DTMF 5

[6] 2008/12/25 11:23:35: Received DTMF 5

[6] 2008/12/25 11:23:38: Received DTMF #

[9] 2008/12/25 11:23:38: Dialplan: Evaluating !^(9411)@.*!sip:18005558355@\r;user=phone!i against 7184365555@localhost

[9] 2008/12/25 11:23:38: Dialplan: Evaluating !^([0-9]{7})@.*!sip:\1@\r;user=phone!i against 7184365555@localhost

[9] 2008/12/25 11:23:38: Dialplan: Evaluating !^([0-9]{10})@.*!sip:1\1@\r;user=phone!i against 7184365555@localhost

[5] 2008/12/25 11:23:38: Dialplan outbound: Match 7184365555@localhost to <sip:17184365555@outbound1.vitelity.net;user=phone> on trunk vitelity outbound

[8] 2008/12/25 11:23:38: Play audio_moh/noise.wav

[9] 2008/12/25 11:23:38: UDP: Opening socket on port 62514

[9] 2008/12/25 11:23:38: UDP: Opening socket on port 62515

[9] 2008/12/25 11:23:38: Resolve 50150: url sip:outbound1.vitelity.net

[9] 2008/12/25 11:23:38: Resolve 50150: naptr outbound1.vitelity.net

[8] 2008/12/25 11:23:38: DNS: Add dns_naptr outbound1.vitelity.net (ttl=60)

[9] 2008/12/25 11:23:38: Resolve 50150: naptr outbound1.vitelity.net

[9] 2008/12/25 11:23:38: Resolve 50150: srv tls _sips._tcp.outbound1.vitelity.net

[8] 2008/12/25 11:23:38: DNS: Add dns_srv _sips._tcp.outbound1.vitelity.net (ttl=60)

[9] 2008/12/25 11:23:38: Resolve 50150: srv tls _sips._tcp.outbound1.vitelity.net

[9] 2008/12/25 11:23:38: Resolve 50150: srv tcp _sip._tcp.outbound1.vitelity.net

[8] 2008/12/25 11:23:38: DNS: Add dns_srv _sip._tcp.outbound1.vitelity.net (ttl=60)

[9] 2008/12/25 11:23:38: Resolve 50150: srv tcp _sip._tcp.outbound1.vitelity.net

[9] 2008/12/25 11:23:38: Resolve 50150: srv udp _sip._udp.outbound1.vitelity.net

[7] 2008/12/25 11:23:38: SIP Rx udp:66.114.64.65:2048:

REGISTER sip:sip.solutions.com SIP/2.0

Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-g6b5c7lxejtn;rport

From: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=hy6e1yqunm

To: "Moishe Grunstein" <sip:401@sip.solutions.com>

Call-ID: 3c26701b6b59-zzy3nchymw3e

CSeq: 5023 REGISTER

Max-Forwards: 70

Contact: <sip:401@192.168.81.102:2048;line=7uuu69wp>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:7035dcbf-7d47-44e4-9f8a-06b3fb57c1b6>"

Contact: <http://192.168.81.102:80>

Contact: <https://192.168.81.102:443>

User-Agent: snom360/7.3.10a

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.81.102

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[9] 2008/12/25 11:23:38: Resolve 50151: udp 66.114.64.65 2048

[7] 2008/12/25 11:23:38: SIP Tx udp:66.114.64.65:2048:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-g6b5c7lxejtn;rport=2048;received=66.114.64.65

From: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=hy6e1yqunm

To: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=673192d660

Call-ID: 3c26701b6b59-zzy3nchymw3e

CSeq: 5023 REGISTER

Contact: <sip:401@192.168.81.102:2048;line=7uuu69wp>;expires=32

Contact: <http://192.168.81.102:80>;expires=32

Contact: <https://192.168.81.102:443>;expires=32

Content-Length: 0

 

 

[8] 2008/12/25 11:23:38: DNS: Add dns_srv _sip._udp.outbound1.vitelity.net (ttl=60)

[9] 2008/12/25 11:23:38: Resolve 50150: srv udp _sip._udp.outbound1.vitelity.net

[9] 2008/12/25 11:23:38: Resolve 50150: a udp outbound1.vitelity.net 5060

[8] 2008/12/25 11:23:38: DNS: Add dns_a outbound1.vitelity.net 64.2.142.87 (ttl=864)

[9] 2008/12/25 11:23:38: Resolve 50150: a udp outbound1.vitelity.net 5060

[9] 2008/12/25 11:23:38: Resolve 50150: udp 64.2.142.86 5060

[7] 2008/12/25 11:23:38: SIP Tx udp:64.2.142.86:5060:

INVITE sip:17184365555@outbound1.vitelity.net;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK-cc1c0f10a8de9fd8f30569e582ffce5f;rport

From: "9175551212" <sip:9175551212@localhost;user=phone>;tag=58440

To: <sip:17184365555@outbound1.vitelity.net;user=phone>

Call-ID: 60d6e30b@pbx

CSeq: 23051 INVITE

Max-Forwards: 70

Contact: <sip:torn_pbx@192.168.10.2:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.1.1.3113

P-Asserted-Identity: <sip:torn_pbx@outbound1.vitelity.net>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 9354 9354 IN IP4 192.168.10.2

s=-

c=IN IP4 192.168.10.2

t=0 0

m=audio 62514 RTP/AVP 0 8 9 18 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/12/25 11:23:38: SIP Rx udp:64.2.142.20:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK-cc1c0f10a8de9fd8f30569e582ffce5f;received=216.112.126.83;rport=5060

From: "9175551212" <sip:9175551212@localhost;user=phone>;tag=58440

To: <sip:17184365555@outbound1.vitelity.net;user=phone>

Call-ID: 60d6e30b@pbx

CSeq: 23051 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:17184365555@64.2.142.20>

Content-Length: 0

 

 

[7] 2008/12/25 11:23:39: SIP Rx udp:74.64.70.251:1323:

REGISTER sip:sip.solutions.com SIP/2.0

Via: SIP/2.0/UDP 192.168.0.253:5060;rport;branch=z9hG4bK2126580982

From: <sip:420@sip.solutions.com>;tag=518775785

To: <sip:420@sip.solutions.com>

Call-ID: 801557023@192.168.0.253

CSeq: 919 REGISTER

Contact: <sip:420@192.168.0.253:5060>

Max-Forwards: 5

User-Agent: Linphone-1.1.0 MX-Video/eXosip

Expires: 200

Content-Length: 0

 

 

[9] 2008/12/25 11:23:39: Resolve 50152: udp 74.64.70.251 1323

[7] 2008/12/25 11:23:39: SIP Tx udp:74.64.70.251:1323:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 192.168.0.253:5060;rport=1323;branch=z9hG4bK2126580982;received=74.64.70.251

From: <sip:420@sip.solutions.com>;tag=518775785

To: <sip:420@sip.solutions.com>;tag=b882f81be3

Call-ID: 801557023@192.168.0.253

CSeq: 919 REGISTER

User-Agent: pbxnsip-PBX/3.1.1.3113

WWW-Authenticate: Digest realm="sip.solutions.com",nonce="8313bfe2108fc54eadd0c0ec083443e5",domain="sip:sip.solutions.com",algorithm=MD5

Content-Length: 0

 

 

[7] 2008/12/25 11:23:40: SIP Rx udp:64.2.142.20:5060:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK-cc1c0f10a8de9fd8f30569e582ffce5f;received=216.112.126.83;rport=5060

From: "9175551212" <sip:9175551212@localhost;user=phone>;tag=58440

To: <sip:17184365555@outbound1.vitelity.net;user=phone>;tag=as0bb732e3

Call-ID: 60d6e30b@pbx

CSeq: 23051 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:17184365555@64.2.142.20>

Content-Length: 0

 

 

[8] 2008/12/25 11:23:40: Play audio_en/ringback.wav

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When placing an outbound call using a cell phone, it is still showing the cell phone number as the caller ID, I thought this was supposed to be fixed in this version. 3.1.1.3113 (Win32), I tied with 2 carriers, Broadvox and Vitelity, all other caller ID functions are working correctly, does it have to do with Remote Party/Privacy Indication:? I tried several setting they all show the cell phone number as the caller ID, except no indication, which shows unavailabe.

 

[8] 2008/12/25 11:23:40: Play audio_en/ringback.wav

 

I assume you have checked this page already http://wiki.pbxnsip.com/index.php/Outbound_Calls_on_Trunk

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I assume you have checked this page already http://wiki.pbxnsip.com/index.php/Outbound_Calls_on_Trunk

As I stated above caller ID is working correctly, my problem is that the caller ID issue http://wiki.pbxnsip.com/index.php/Release_..._3.1#Cell_Phone does not seem to be working for me, it still displays the cell phone number as the caller ID.

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