Name showing when dialing extension number

[shopcomputer]

I just set up a client with latest 4.5.1030, using Snom 300 and 720 phones.

 

When they call another extension is shows calling xxx.localhost instead of showing the name of the person they are calling.

 

The directory button pulls up the names fine.

[Vodia support]

Can you post a siptrace of the call?

 

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

[Vodia PBX]

Can you post a siptrace of the call?

 

I guess that will not show the problem. Generally speaking, the phones have a setting on how to show a SIP caller-ID in the form "display name" <sip:user@domain>. Usually not everything fits on the screen, so the phone picks some parts of it, which is controlled by a setting. Probably the problem is that we need to differentiate between the models when provisioning this setting. E.g. on the snom 300, you probably just want to see the user part, while on the devices with large color display, you want to see the name.

[shopcomputer]

I guess that will not show the problem. Generally speaking, the phones have a setting on how to show a SIP caller-ID in the form "display name" <sip:user@domain>. Usually not everything fits on the screen, so the phone picks some parts of it, which is controlled by a setting. Probably the problem is that we need to differentiate between the models when provisioning this setting. E.g. on the snom 300, you probably just want to see the user part, while on the devices with large color display, you want to see the name.

This is not a caller ID issue, this is a ldap lookup issue, problem is with the callers phone not showing the name of the person he is calling.

 

5] 2012/02/16 21:01:55:

 

SIP Rx tls:192.168.2.52:40159:

 

 

 

INVITE sip:200@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.2.52:40159;branch=z9hG4bK-caq8414m71p8;rport

From: "Ronald M Weiss" <sip:201@localhost>;tag=qusj28c34z

To: <sip:200@localhost;user=phone>

Call-ID: 18b53d4f5033-hlbw26u4efrp

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:201@192.168.2.52:40159;transport=tls;line=m6u5wcbu>;reg-id=1

X-Serialnumber: 0004137004E1

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

User-Agent: snom720/8.7.3.201202132240

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: 735

 

v=0

o=root 1945276751 1945276751 IN IP4 192.168.2.52

s=call

c=IN IP4 192.168.2.52

t=0 0

m=audio 53042 RTP/AVP 9 0 8 3 97 98 99 100 106 107 108 109 18 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:4FFC8dTY4Aics1WC48YzDKqCKTvx7aXnE8kFqsGi

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 G726-16/8000

a=rtpmap:98 G726-24/8000

a=rtpmap:99 G726-32/8000

a=rtpmap:100 G726-40/8000

a=rtpmap:106 AAL2-G726-16/8000

a=rtpmap:107 AAL2-G726-24/8000

a=rtpmap:108 AAL2-G726-32/8000

a=rtpmap:109 AAL2-G726-40/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

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

a=sendrecv

 

 

 

 

[8] 2012/02/16 21:01:55:

 

Allocating for call port 127, SIP call id 18b53d4f5033-hlbw26u4efrp

 

 

 

[8] 2012/02/16 21:01:55:

 

Could not find a trunk (3 trunks)

 

 

 

[5] 2012/02/16 21:01:55:

 

SIP Tx tls:192.168.2.52:40159:

 

 

 

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.2.52:40159;branch=z9hG4bK-caq8414m71p8;rport=40159

From: "Ronald M Weiss" <sip:201@localhost>;tag=qusj28c34z

To: <sip:200@localhost;user=phone>;tag=f9b616cc8b

Call-ID: 18b53d4f5033-hlbw26u4efrp

CSeq: 1 INVITE

Content-Length: 0

 

 

 

 

 

[7] 2012/02/16 21:01:55:

 

Set packet length to 20

 

 

 

[6] 2012/02/16 21:01:55:

 

Call-leg 127: Sending RTP for 18b53d4f5033-hlbw26u4efrp to 192.168.2.52:53042, codec not set yet

 

 

 

[8] 2012/02/16 21:01:55:

 

Call from an user 201

 

 

 

[8] 2012/02/16 21:01:55:

 

To is <sip:200@localhost;user=phone>, user 23, domain 1

 

 

 

[8] 2012/02/16 21:01:55:

 

To user 200

 

 

 

[8] 2012/02/16 21:01:55:

 

Call state for call object 58: idle

 

 

 

[8] 2012/02/16 21:01:55:

 

Call state for call object 58: alerting

 

 

 

[8] 2012/02/16 21:01:55:

 

Play audio_moh/noise.wav, caching true

 

 

 

[7] 2012/02/16 21:01:55:

 

Call port 127: set_codecs for 18b53d4f5033-hlbw26u4efrp codecs "", codec_preference count 7

 

 

 

[8] 2012/02/16 21:01:55:

 

Allocating for call port 128, SIP call id ebc83077@pbx

 

 

 

[7] 2012/02/16 21:01:55:

 

Call port 128: set_codecs for ebc83077@pbx codecs "", codec_preference count 7

 

 

 

[5] 2012/02/16 21:01:55:

 

SIP Tx tls:192.168.2.40:2072:

[shopcomputer]

I can't figure this out, on my in house demo system if I call another extension it shows the name of the person I am calling, on my customers system with same 4.5 and same 720 phone it shows calling 200@localhost?