SnomOne 4 outgoing identity hidden

[eyeless]

Hi,

 

We upgraded to the latest version of SnomOne 4: Zeta Perseids (4.5.1.1107) (Mac OS X)

Apart from having to upgrade the Snom M9 telephones in order to get two-way voice, we only have one problem left and that is that all outgoing calls are now displayed as hidden to the recipient and I cannot simply find a way to make the recipient see what number the calls are coming from any longer.

Anyone who has a hint of what might have changed from version 4.2-4.3 to the 4.5 version that could affect this?

 

All the best,

Jerry

[Vodia PBX]

Yea, that's a "classic" for the upgrade to 4.5 and higher. That was a side effect of having more control over the SIP headers. You need to change the header presentation in the trunk. There are actually some posts on the forum about this. I would suggest to try the few standard cases out which are available in the drop down (Remote-Party-ID, RFC3325, ...); usually one of them works with your service provider.

[eyeless]

Yea, that's a "classic" for the upgrade to 4.5 and higher. That was a side effect of having more control over the SIP headers. You need to change the header presentation in the trunk. There are actually some posts on the forum about this. I would suggest to try the few standard cases out which are available in the drop down (Remote-Party-ID, RFC3325, ...); usually one of them works with your service provider.

 

Yes, I saw that field, but was not sure if it would be meaningful to change it … will see what I can do to get it to work …. .

[eyeless]

Well, the provider did not understand what was wrong or how to change to a custom header. I tried all the different alternatives in the Number/Call Identification section now (well, almost all, at least all in the drop-down menu).

 

Here is the log if it makes anything more clear (I only have added to the Trunk ANI field the phone number after upgrade - this field was blank before, but adding the number there did not change anything) - the hidden outgoing number is 031109430 (account name is both 30 & 031109430):

 

[5] 2013/02/08 16:11:29: SIP Rx tls:10.0.3.234:3448:

INVITE sip:0317018939@10.0.3.10;user=phone SIP/2.0

Via: SIP/2.0/TLS 10.0.3.234:3448;branch=z9hG4bK-hcuffvg1f2s0;rport

From: "Eva Levin" <sip:30@10.0.3.10>;tag=s1609stbfc

To: <sip:0317018939@10.0.3.10;user=phone>

Call-ID: 3c31d1beece8-lhkffaqe75rn

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:30@10.0.3.234:3448;transport=tls;line=omt9jyts>;reg-id=1

X-Serialnumber: 00041331E1F5

P-Key-Flags: keys="3"

User-Agent: snom320/8.4.18

Accept: application/sdp

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

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

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 518

 

v=0

o=root 484433817 484433817 IN IP4 10.0.3.234

s=call

c=IN IP4 10.0.3.234

t=0 0

m=audio 64710 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:E2SAYAZBeodnsVnNhmM1XR7Y9uOpVl68nShKlNlx

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] 2013/02/08 16:11:29: Packet authenticated by transport layer

[9] 2013/02/08 16:11:29: Using outbound proxy sip:10.0.3.234:3448;transport=tls because of flow-label

[9] 2013/02/08 16:11:30: Last message repeated 3 times

[5] 2013/02/08 16:11:30: SIP Tx tls:10.0.3.234:3448:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 10.0.3.234:3448;branch=z9hG4bK-hcuffvg1f2s0;rport=3448

From: "Eva Levin" <sip:30@10.0.3.10>;tag=s1609stbfc

To: <sip:0317018939@10.0.3.10;user=phone>;tag=8d078c94b0

Call-ID: 3c31d1beece8-lhkffaqe75rn

CSeq: 1 INVITE

Content-Length: 0

 

[8] 2013/02/08 16:11:30: Incoming call: Request URI sip:0317018939@10.0.3.10;user=phone, To is <sip:0317018939@10.0.3.10;user=phone>

[8] 2013/02/08 16:11:30: Set the To domain based on From user 30@10.0.3.10

[9] 2013/02/08 16:11:30: Resolve 10167: url sip:sip.voicetech.se

[9] 2013/02/08 16:11:30: Resolve 10167: naptr sip.voicetech.se

[9] 2013/02/08 16:11:30: Resolve 10167: srv tls _sips._tcp.sip.voicetech.se

[9] 2013/02/08 16:11:30: Resolve 10167: srv tcp _sip._tcp.sip.voicetech.se

[9] 2013/02/08 16:11:30: Resolve 10167: srv udp _sip._udp.sip.voicetech.se

[9] 2013/02/08 16:11:30: Resolve 10167: aaaa udp sip.voicetech.se 5060

[9] 2013/02/08 16:11:30: Resolve 10167: a udp sip.voicetech.se 5060

[9] 2013/02/08 16:11:30: Resolve 10167: udp 212.3.0.180 5060

[5] 2013/02/08 16:11:30: SIP Tx udp:212.3.0.180:5060:

INVITE sip:0317018939@sip.voicetech.se;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-d480b58ea53b1be72eb8bf65a62c4c8a;rport

From: "Eva Levin" <sip:031109430@10.0.3.10;user=phone>;tag=879293534

To: <sip:0317018939@10.0.3.10;user=phone>

Call-ID: 173c1240@pbx

CSeq: 25139 INVITE

Max-Forwards: 70

Contact: <sip:031109430@10.0.3.10:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/4.5.1.1107 Zeta Perseids

P-Asserted-Identity: "Eva Levin" <sip:031109430@sip.voicetech.se>

Privacy: id

Content-Type: application/sdp

Content-Length: 378

 

v=0

o=- 1818723674 1818723674 IN IP4 10.0.3.10

s=-

c=IN IP4 10.0.3.10

t=0 0

m=audio 50058 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] 2013/02/08 16:11:30: set codec: codec pcmu/8000 is set to call-leg 139

[5] 2013/02/08 16:11:30: SIP Tx tls:10.0.3.234:3448:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TLS 10.0.3.234:3448;branch=z9hG4bK-hcuffvg1f2s0;rport=3448

From: "Eva Levin" <sip:30@10.0.3.10>;tag=s1609stbfc

To: <sip:0317018939@10.0.3.10;user=phone>;tag=8d078c94b0

Call-ID: 3c31d1beece8-lhkffaqe75rn

CSeq: 1 INVITE

Contact: <sip:30@10.0.3.10:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/4.5.1.1107 Zeta Perseids

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 474

 

v=0

o=- 1414671445 1414671445 IN IP4 10.0.3.10

s=-

c=IN IP4 10.0.3.10

t=0 0

m=audio 58816 RTP/AVP 0 8 9 18 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ssWlPg8UG7cr5hZuHpunzJ9JfCBJFmfdgZAPsbPa

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=ptime:20

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

a=sendrecv

[5] 2013/02/08 16:11:30: SIP Rx udp:212.3.0.180:5060:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-d480b58ea53b1be72eb8bf65a62c4c8a;rport=47104

From: "Eva Levin" <sip:031109430@10.0.3.10:5060;user=phone>;tag=879293534

To: <sip:0317018939@10.0.3.10:5060;user=phone>;tag=7000166881dd2f9a2a8458b004d02617.ff08

Call-ID: 173c1240@pbx

CSeq: 25139 INVITE

Proxy-Authenticate: Digest realm="sips.teleman.com", nonce="511519265cc9bda9b08f820a70c78daa8fc448af"

Server: OpenSer (1.1.0-tls (x86_64/linux))

Content-Length: 0

Warning: 392 212.3.0.180:5060 "Noisy feedback tells: pid=29552 req_src_ip=81.216.208.134 req_src_port=47104 in_uri=sip:0317018939@sip.voicetech.se;user=phone out_uri=sip:0317018939@sip.voicetech.se;user=phone via_cnt==1"

 

[8] 2013/02/08 16:11:30: Answer challenge with username 031109430

[9] 2013/02/08 16:11:30: Resolve 10169: udp 212.3.0.180 5060 udp:1

[5] 2013/02/08 16:11:30: SIP Tx udp:212.3.0.180:5060:

ACK sip:0317018939@sip.voicetech.se;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-d480b58ea53b1be72eb8bf65a62c4c8a;rport

From: "Eva Levin" <sip:031109430@10.0.3.10:5060;user=phone>;tag=879293534

To: <sip:0317018939@10.0.3.10:5060;user=phone>;tag=7000166881dd2f9a2a8458b004d02617.ff08

Call-ID: 173c1240@pbx

CSeq: 25139 ACK

Max-Forwards: 70

Content-Length: 0

 

[9] 2013/02/08 16:11:30: Resolve 10170: udp 212.3.0.180 5060 udp:1

[5] 2013/02/08 16:11:30: SIP Tx udp:212.3.0.180:5060:

INVITE sip:0317018939@sip.voicetech.se;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-60034bf93976763369c60cb96a63233e;rport

From: "Eva Levin" <sip:031109430@10.0.3.10;user=phone>;tag=879293534

To: <sip:0317018939@10.0.3.10;user=phone>

Call-ID: 173c1240@pbx

CSeq: 25140 INVITE

Max-Forwards: 70

Contact: <sip:031109430@10.0.3.10:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snomONE/4.5.1.1107 Zeta Perseids

P-Asserted-Identity: "Eva Levin" <sip:031109430@sip.voicetech.se>

Privacy: id

Proxy-Authorization: Digest realm="sips.teleman.com",nonce="511519265cc9bda9b08f820a70c78daa8fc448af",response="b49cb9031c05e2a49124edc6790d170a",username="031109430",uri="sip:0317018939@sip.voicetech.se;user=phone",algorithm=MD5

Content-Type: application/sdp

Content-Length: 378

 

v=0

o=- 1818723674 1818723674 IN IP4 10.0.3.10

s=-

c=IN IP4 10.0.3.10

t=0 0

m=audio 50058 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

[9] 2013/02/08 16:11:30: Message repetition, packet dropped

[5] 2013/02/08 16:11:30: SIP Rx udp:212.3.0.180:5060:

SIP/2.0 100 trying -- your call is important to us

Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-60034bf93976763369c60cb96a63233e;rport=47104

From: "Eva Levin" <sip:031109430@10.0.3.10:5060;user=phone>;tag=879293534

To: <sip:0317018939@10.0.3.10:5060;user=phone>

Call-ID: 173c1240@pbx

CSeq: 25140 INVITE

Server: OpenSer (1.1.0-tls (x86_64/linux))

Content-Length: 0

Warning: 392 212.3.0.180:5060 "Noisy feedback tells: pid=29553 req_src_ip=81.216.208.134 req_src_port=47104 in_uri=sip:0317018939@sip.voicetech.se;user=phone out_uri=sip:0317018939@sip4.teleman.com;user=phone via_cnt==1"

 

[9] 2013/02/08 16:11:30: Message repetition, packet dropped

[5] 2013/02/08 16:11:30: SIP Rx tls:10.0.3.234:3448:

PRACK sip:30@10.0.3.10:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 10.0.3.234:3448;branch=z9hG4bK-oy0pueeg4a5m;rport

From: "Eva Levin" <sip:30@10.0.3.10>;tag=s1609stbfc

To: <sip:0317018939@10.0.3.10;user=phone>;tag=8d078c94b0

Call-ID: 3c31d1beece8-lhkffaqe75rn

CSeq: 2 PRACK

Max-Forwards: 70

Contact: <sip:30@10.0.3.234:3448;transport=tls;line=omt9jyts>;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] 2013/02/08 16:11:30: Packet authenticated by transport layer

[5] 2013/02/08 16:11:30: SIP Tx tls:10.0.3.234:3448:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.0.3.234:3448;branch=z9hG4bK-oy0pueeg4a5m;rport=3448

From: "Eva Levin" <sip:30@10.0.3.10>;tag=s1609stbfc

To: <sip:0317018939@10.0.3.10;user=phone>;tag=8d078c94b0

Call-ID: 3c31d1beece8-lhkffaqe75rn

CSeq: 2 PRACK

Contact: <sip:30@10.0.3.10:5061;transport=tls>

User-Agent: snomONE/4.5.1.1107 Zeta Perseids

Content-Length: 0

 

[5] 2013/02/08 16:11:30: SIP Rx udp:212.3.0.180:5060:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-60034bf93976763369c60cb96a63233e;rport=47104

Record-Route: <sip:212.3.0.180;lr=on;ftag=879293534>

Contact: <sip:0317018939@212.3.0.165:5060;transport=udp>

To: <sip:0317018939@10.0.3.10:5060;user=phone>;tag=97be3903

From: "Eva Levin"<sip:031109430@10.0.3.10:5060;user=phone>;tag=879293534

Call-ID: 173c1240@pbx

CSeq: 25140 INVITE

Content-Type: application/sdp

Content-Length: 367

 

v=0

o=- 19337546 0 IN IP4 88.131.198.35

s=Cisco SDP 0

c=IN IP4 62.80.216.14

t=0 0

m=audio 43684 RTP/AVP 8 101 100

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:100 X-NSE/8000

a=fmtp:100 192-194,200-202

a=X-sqn:0

a=X-cap: 1 audio RTP/AVP 100

a=X-cpar: a=rtpmap:100 X-NSE/8000

a=X-cpar: a=fmtp:100 192-194,200-202

a=X-cap: 2 image udptl t38

[5] 2013/02/08 16:11:30: set codec: codec pcma/8000 is set to call-leg 140

[Vodia support]

If you're trying to just advertise the trunk ANI you can do something like this..

Here are more examples you can actually get real fancy by using the custom header.

 

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

 

 

[eyeless]

If you're trying to just advertise the trunk ANI you can do something like this..

Here are more examples you can actually get real fancy by using the custom header.

 

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

 

 

 

I see. Not very helpful though as I only want the SnomOne to send out headers just like it always has done up to version 4.5.x … when there has been no problem. Apparently the trunk providers are using Asterix servers themselves and do not understand SnomOne … . So was the reason for changing this to get people to buy the SnomOne 5 instead? One wonders … .

[Vodia PBX]

I see. Not very helpful though as I only want the SnomOne to send out headers just like it always has done up to version 4.5.x … when there has been no problem. Apparently the trunk providers are using Asterix servers themselves and do not understand SnomOne … . So was the reason for changing this to get people to buy the SnomOne 5 instead? One wonders … .

 

The reason was that practically every service provider out there reads the RFC in a different way and a simple drop-down was not enough to cover the cases. Most service providers assume(d) that you are registering a soft-phone with exactly one phone number associated, and when it comes to cell phone inclusion and you want to see the phone number of the one who actually called in, things get very messy.

[eyeless]

The reason was that practically every service provider out there reads the RFC in a different way and a simple drop-down was not enough to cover the cases. Most service providers assume(d) that you are registering a soft-phone with exactly one phone number associated, and when it comes to cell phone inclusion and you want to see the phone number of the one who actually called in, things get very messy.

 

I understand. I will try and work this out with the trunk provider, and it should in their interest to be able to help out in such cases, so … . Will post the solution here if I get one.

[Vodia PBX]

So it seems that for voicetech.se the following settings are suitable:

• Remote Party/Privacy Indication: Custom Headers
• Request-URI: Let the system decide
• From: Trunk ANI
• To: Let the system decide
• P-Asserted-Identity: Based on trunk account info
• P-Preferred-Identity: Don't use header
• Remote-Party-ID: Don't use header
• P-Charging-Vector: Don't use header
• Privacy Indication: Don't use header

Then you need to put your phone number into the setting Trunk ANI of the PBX.

[eyeless]

So it seems that for voicetech.se the following settings are suitable:

• Remote Party/Privacy Indication: Custom Headers
• Request-URI: Let the system decide
• From: Trunk ANI
• To: Let the system decide
• P-Asserted-Identity: Based on trunk account info
• P-Preferred-Identity: Don't use header
• Remote-Party-ID: Don't use header
• P-Charging-Vector: Don't use header
• Privacy Indication: Don't use header

Then you need to put your phone number into the setting Trunk ANI of the PBX.

 

Yes, but the last: "Then you need to put your phone number into the setting Trunk ANI of the PBX." might not be needed as I received the exact same result with it filled in or not.

Thanks!!