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Sangoma B600D on Snom One outgoing calls always busy

Obed Alba

By Obed Alba
in General Setup

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Obed Alba Newbie

Obed Alba

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Hello All

I have a System running good for a while, but recently seems so weird, My setup is a Sangoma B600D with Snom One Unlimited and for some reason can receive calls, but can not make outoging calls it always sound busy,internal calls between extensions works fine in both ways.

 

Any ideas what might be causing this, or where to look for clues?

 

Thanks.

 

This is my log

[9] 2013/02/28 15:34:23: Resolve 1682: aaaa udp 192.168.137.108 5060

[9] 2013/02/28 15:34:23: Resolve 1682: a udp 192.168.137.108 5060

[9] 2013/02/28 15:34:23: Resolve 1682: udp 192.168.137.108 5060

[5] 2013/02/28 15:34:25: SIP Rx udp:192.168.137.154:2048:

INVITE sip:7261247@192.168.137.254;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-em3z2eypz8y4;rport

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:208@192.168.137.154:2048;line=x83vrg9s>;reg-id=1

X-Serialnumber: 00041335C884

P-Key-Flags: keys="3"

User-Agent: snom320/8.4.35

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

Call-Info: <sip:192.168.137.254>;appearance-index=1

Session-Expires: 3600;refresher=uas

Min-SE: 90

Content-Type: application/sdp

Content-Length: 481

 

v=0

o=root 1638961721 1638961721 IN IP4 192.168.137.154

s=call

c=IN IP4 192.168.137.154

t=0 0

m=audio 56492 RTP/AVP 0 8 9 99 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:hYBafrqHq8h5F5E2TNCH8xfBRVgk8N57f86tv9mf

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:99 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=sendrecv

[8] 2013/02/28 15:34:25: Allocating for call port 122, SIP call id 3c269177ceb5-a1za6z3kewa1

[9] 2013/02/28 15:34:25: UDP(IPv4): Opening socket on 0.0.0.0:53328

[9] 2013/02/28 15:34:25: UDP(IPv4): Opening socket on 0.0.0.0:53329

[8] 2013/02/28 15:34:25: Could not find a trunk (1 trunks)

[9] 2013/02/28 15:34:25: Resolve 1683: aaaa udp 192.168.137.154 2048

[9] 2013/02/28 15:34:25: Resolve 1683: a udp 192.168.137.154 2048

[9] 2013/02/28 15:34:25: Resolve 1683: udp 192.168.137.154 2048

[5] 2013/02/28 15:34:25: SIP Tx udp:192.168.137.154:2048:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-em3z2eypz8y4;rport=2048

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 1 INVITE

Content-Length: 0

[9] 2013/02/28 15:34:25: Resolve 1684: aaaa udp 192.168.137.154 2048

[9] 2013/02/28 15:34:25: Resolve 1684: a udp 192.168.137.154 2048

[9] 2013/02/28 15:34:25: Resolve 1684: udp 192.168.137.154 2048

[5] 2013/02/28 15:34:25: SIP Tx udp:192.168.137.154:2048:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-em3z2eypz8y4;rport=2048

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 1 INVITE

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

WWW-Authenticate: Digest realm="192.168.137.254",nonce="8a3ad8f16e25a19e70270697b124c7d4",domain="sip:7261247@192.168.137.254;user=phone",algorithm=MD5

Content-Length: 0

[5] 2013/02/28 15:34:26: SIP Rx udp:192.168.137.154:2048:

ACK sip:7261247@192.168.137.254;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-em3z2eypz8y4;rport

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:208@192.168.137.154:2048;line=x83vrg9s>;reg-id=1

Content-Length: 0

[5] 2013/02/28 15:34:26: SIP Rx udp:192.168.137.154:2048:

INVITE sip:7261247@192.168.137.254;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-crzvtm8g5izl;rport

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 2 INVITE

Max-Forwards: 70

Contact: <sip:208@192.168.137.154:2048;line=x83vrg9s>;reg-id=1

X-Serialnumber: 00041335C884

P-Key-Flags: keys="3"

User-Agent: snom320/8.4.35

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

Call-Info: <sip:192.168.137.254>;appearance-index=1

Session-Expires: 3600;refresher=uas

Min-SE: 90

Authorization: Digest username="208",realm="192.168.137.254",nonce="8a3ad8f16e25a19e70270697b124c7d4",uri="sip:7261247@192.168.137.254;user=phone",response="5d37764e2b30417637b84556437b2258",algorithm=MD5

Content-Type: application/sdp

Content-Length: 481

 

v=0

o=root 1638961721 1638961721 IN IP4 192.168.137.154

s=call

c=IN IP4 192.168.137.154

t=0 0

m=audio 56492 RTP/AVP 0 8 9 99 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:hYBafrqHq8h5F5E2TNCH8xfBRVgk8N57f86tv9mf

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:99 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=sendrecv

[8] 2013/02/28 15:34:26: Tagging request with existing tag

[7] 2013/02/28 15:34:26: Set packet length to 20

[6] 2013/02/28 15:34:26: Call-leg 122: Sending RTP for 3c269177ceb5-a1za6z3kewa1 to 192.168.137.154:56492, codec not set yet

[8] 2013/02/28 15:34:26: Incoming call: Request URI sip:7261247@192.168.137.254;user=phone, To is <sip:7261247@192.168.137.254;user=phone>

[8] 2013/02/28 15:34:26: Call from an user 208

[9] 2013/02/28 15:34:26: Resolve 1685: aaaa udp 192.168.137.154 2048

[9] 2013/02/28 15:34:26: Resolve 1685: a udp 192.168.137.154 2048

[9] 2013/02/28 15:34:26: Resolve 1685: udp 192.168.137.154 2048

[5] 2013/02/28 15:34:26: SIP Tx udp:192.168.137.154:2048:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-crzvtm8g5izl;rport=2048

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 2 INVITE

Content-Length: 0

[8] 2013/02/28 15:34:26: To is <sip:7261247@192.168.137.254;user=phone>, user 0, domain 1

[8] 2013/02/28 15:34:26: From user 208

[8] 2013/02/28 15:34:26: Set the To domain based on From user 208@pbx.company.com

[8] 2013/02/28 15:34:26: Call state for call object 57: idle

[7] 2013/02/28 15:34:26: Call port 122: set_codecs for 3c269177ceb5-a1za6z3kewa1 codecs "", codec_preference count 7

[9] 2013/02/28 15:34:26: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 7261247@192.168.137.254

[5] 2013/02/28 15:34:26: Dialplan "Standar": Match 7261247@192.168.137.254 to sip:7261247@please.change;user=phone on trunk PSTN 8FXO

[9] 2013/02/28 15:34:26: Generating ht header using {to}

[9] 2013/02/28 15:34:26: Generating hpai header using {trunk}

[8] 2013/02/28 15:34:26: Play audio_moh/noise.wav, caching true

[8] 2013/02/28 15:34:26: Allocating for call port 123, SIP call id 10fe8e01@pbx

[9] 2013/02/28 15:34:26: UDP(IPv4): Opening socket on 0.0.0.0:55444

[9] 2013/02/28 15:34:26: UDP(IPv4): Opening socket on 0.0.0.0:55445

[7] 2013/02/28 15:34:26: Call port 123: set_codecs for 10fe8e01@pbx codecs "", codec_preference count 7

[8] 2013/02/28 15:34:26: call port 123: state code from 0 to 100

[9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec pcmu/8000 to available list

[9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec pcma/8000 to available list

[9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec g722/8000 to available list

[9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec g729/8000 to available list

[9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec g726-32/8000 to available list

[9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec gsm/8000 to available list

[9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: codec_preference size 7, available codecs size 7

[9] 2013/02/28 15:34:26: Resolve 1686: url sip:192.168.137.254:5066

[9] 2013/02/28 15:34:26: Resolve 1686: udp 192.168.137.254 5066

[5] 2013/02/28 15:34:26: SIP Tx udp:192.168.137.254:5066:

INVITE sip:7261247@please.change;user=phone SIP/2.0

Via: SIP/2.0/UDP 216.150.32.145:5060;branch=z9hG4bK-0e14ba8c5e81c99046bac1b75c04418a;rport

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=646572893

To: <sip:7261247@pbx.company.com;user=phone>

Call-ID: 10fe8e01@pbx

CSeq: 98 INVITE

Max-Forwards: 70

Contact: <sip:anonymous@216.150.32.145:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

P-Asserted-Identity: <sip:please.change>

Privacy: id

Content-Type: application/sdp

Content-Length: 386

 

v=0

o=- 672703835 672703835 IN IP4 216.150.32.145

s=-

c=IN IP4 216.150.32.145

t=0 0

m=audio 55444 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=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[5] 2013/02/28 15:34:26: SIP Rx udp:192.168.137.254:5066:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 216.150.32.145:5060;branch=z9hG4bK-0e14ba8c5e81c99046bac1b75c04418a;rport=5060;received=192.168.137.254

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=646572893

To: <sip:7261247@pbx.company.com;user=phone>;tag=ds-35dee0d-9e9b8a88

Call-ID: 10fe8e01@pbx

CSeq: 98 INVITE

Content-Length: 0

Server: Netborder Express Gateway/4.3.3

Contact: <sip:NetborderExpressGateway@192.168.137.254:5066;transport=udp>

[9] 2013/02/28 15:34:26: Message repetition, packet dropped

[8] 2013/02/28 15:34:26: call port 122: state code from 0 to 183

[7] 2013/02/28 15:34:26: Set packet length to 20

[9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec pcmu/8000 to available list

[9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec pcma/8000 to available list

[9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec g722/8000 to available list

[9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec g729/8000 to available list

[9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec g726-32/8000 to available list

[9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec gsm/8000 to available list

[9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: codec_preference size 7, available codecs size 7

[6] 2013/02/28 15:34:26: Call-leg 122: Codec pcmu/8000 is chosen for call id 3c269177ceb5-a1za6z3kewa1

[5] 2013/02/28 15:34:26: set codec: codec pcmu/8000 is set to call-leg 122

[9] 2013/02/28 15:34:26: Resolve 1687: aaaa udp 192.168.137.154 2048

[9] 2013/02/28 15:34:26: Resolve 1687: a udp 192.168.137.154 2048

[9] 2013/02/28 15:34:26: Resolve 1687: udp 192.168.137.154 2048

[5] 2013/02/28 15:34:26: SIP Tx udp:192.168.137.154:2048:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-crzvtm8g5izl;rport=2048

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 2 INVITE

Contact: <sip:208@216.150.32.145:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 400

 

v=0

o=- 428979860 428979860 IN IP4 216.150.32.145

s=-

c=IN IP4 216.150.32.145

t=0 0

m=audio 53328 RTP/AVP 0 8 9 18 99 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:99 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] 2013/02/28 15:34:26: SIP Rx udp:216.150.32.145:44407:

PRACK sip:208@216.150.32.145:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-uwp1uuf0hy3v;rport

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 3 PRACK

Max-Forwards: 70

Contact: <sip:208@192.168.137.154:2048;line=x83vrg9s>;reg-id=1

RAck: 1 2 INVITE

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

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

Content-Length: 0

[9] 2013/02/28 15:34:26: Resolve 1688: udp 216.150.32.145 44407

[5] 2013/02/28 15:34:26: SIP Tx udp:216.150.32.145:44407:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-uwp1uuf0hy3v;rport=44407;received=216.150.32.145

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 3 PRACK

Contact: <sip:208@216.150.32.145:5060>

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

Content-Length: 0

[6] 2013/02/28 15:34:26: Call-leg 122: Sending RTP for 3c269177ceb5-a1za6z3kewa1 to 216.150.32.145:30126, codec pcmu/8000

[5] 2013/02/28 15:34:26: SIP Rx udp:192.168.137.254:5066:

SIP/2.0 486 Busy Here

Via: SIP/2.0/UDP 216.150.32.145:5060;branch=z9hG4bK-0e14ba8c5e81c99046bac1b75c04418a;rport=5060;received=192.168.137.254

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=646572893

To: <sip:7261247@pbx.company.com;user=phone>;tag=ds-35dee0d-9e9b8a88

Call-ID: 10fe8e01@pbx

CSeq: 98 INVITE

Content-Length: 0

Server: Netborder Express Gateway/4.3.3

CPD-Result: Busy

Contact: <sip:192.168.137.254:5066;transport=udp>

[7] 2013/02/28 15:34:26: Call 10fe8e01@pbx: Clear last INVITE

[9] 2013/02/28 15:34:26: Resolve 1689: url sip:192.168.137.254:5066

[9] 2013/02/28 15:34:26: Resolve 1689: udp 192.168.137.254 5066

[5] 2013/02/28 15:34:26: SIP Tx udp:192.168.137.254:5066:

ACK sip:7261247@please.change;user=phone SIP/2.0

Via: SIP/2.0/UDP 216.150.32.145:5060;branch=z9hG4bK-0e14ba8c5e81c99046bac1b75c04418a;rport

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=646572893

To: <sip:7261247@pbx.company.com;user=phone>;tag=ds-35dee0d-9e9b8a88

Call-ID: 10fe8e01@pbx

CSeq: 98 ACK

Max-Forwards: 70

Contact: <sip:anonymous@216.150.32.145:5060;transport=udp>

P-Asserted-Identity: <sip:please.change>

Privacy: id

Content-Length: 0

[5] 2013/02/28 15:34:26: INVITE Response 486 Busy Here: Terminate 10fe8e01@pbx

[8] 2013/02/28 15:34:26: Clearing call port 123, SIP call id 10fe8e01@pbx

[8] 2013/02/28 15:34:26: call port 122: state code from 183 to 486

[9] 2013/02/28 15:34:26: Resolve 1690: aaaa udp 192.168.137.154 2048

[9] 2013/02/28 15:34:26: Resolve 1690: a udp 192.168.137.154 2048

[9] 2013/02/28 15:34:26: Resolve 1690: udp 192.168.137.154 2048

[5] 2013/02/28 15:34:26: SIP Tx udp:192.168.137.154:2048:

SIP/2.0 486 Busy Here

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-crzvtm8g5izl;rport=2048

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 2 INVITE

Contact: <sip:208@216.150.32.145:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

Content-Length: 0

[8] 2013/02/28 15:34:26: Remove leg 124: call port 123, SIP call id 10fe8e01@pbx

[8] 2013/02/28 15:34:26: Hangup: Call 123 not found

[5] 2013/02/28 15:34:26: SIP Rx udp:192.168.137.154:2048:

ACK sip:7261247@192.168.137.254;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-crzvtm8g5izl;rport

From: <sip:208@192.168.137.254>;tag=66vco17bd1

To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f

Call-ID: 3c269177ceb5-a1za6z3kewa1

CSeq: 2 ACK

Max-Forwards: 70

Contact: <sip:208@192.168.137.154:2048;line=x83vrg9s>;reg-id=1

Content-Length: 0

[8] 2013/02/28 15:34:26: Clearing call port 122, SIP call id 3c269177ceb5-a1za6z3kewa1

[8] 2013/02/28 15:34:26: Remove leg 123: call port 122, SIP call id 3c269177ceb5-a1za6z3kewa1

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Obed Alba Newbie

Obed Alba

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Check if the user has "Block outgoing caller-ID" set on the extension level.

 

The thing that I noticed in my system is that the snom one can not "communicate" with the sangoma card, for this reason the outbound calls can not be made. I can receive calls, but can not make calls.

 

Any other idea?

 

Thanks

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Vodia support Proficient

Vodia support

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Check the private indication on the trunk level and set it to remote-party ID and all the available setting.

 

http://wiki.snomone.com/index.php?title=Trunks_Number_Call_Identification

 

The Custom header would work as well but you need to know what the provider would like to see in the from field otherwise they will reject it.

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