Name of caller does not appear

asienaert · ‎01-05-2012

I'm currently testing my new SPA508G phone and everything seems to work fine except that the name of the caller is not displayed on incomming calls although the telephone number and name of the caller are in the Personal Directory. The phone seems to recognize the caller since the proper ring tone, the one specified in the personal directory for this user, is used.

Any suggestions?

Thanks,

Alain

nseto · ‎01-05-2012

Options for caller ID is in the SIP tab under SIP Parameters/Caller ID header.

This is dependent on what's being received in the SIP message for the call.

You can obtain this SIP message information by checking the debug log or wireshark trace.

To obtain debug log, the following link can help you.

https://supportforums.cisco.com/docs/DOC-9862

asienaert · ‎01-05-2012

Thanks for your response.

I did try all settings for Caller ID Header but without succes.

I also created a log file but don't know what to do with it .

[CCTRL]record lcr phone=048758980 exten=0 type=0

START_RING in CC_refresh

SDP RTPMAP 101 --> 142

[0:0]AUD ALLOC CALL (port=16522)

[0:0]AUD Rel Call

[CCTRL]record lcr phone=048758980 exten=0

[0]CC:NewCallState 0/10

[CCTRL]record lcr phone=048758980 exten=0 type=0

START_RING in CC_refresh

SDP RTPMAP 101 --> 142

[0:0]AUD ALLOC CALL (port=16522)

[0:0]AUD Rel Call

[CCTRL]record lcr phone=048758980 exten=0

[0]CC:NewCallState 0/10

nseto · ‎01-05-2012

There was an additional link in the link that I gave you that shows you how to turn on sip debugging and to level 3.

Pasted that link here.

https://supportforums.cisco.com/docs/DOC-9934

The sip debug will show the sip invite that shows the name and/or phone number.

asienaert · ‎01-05-2012

OK, got it.

Please let me know if you want me to upload the log file to a server and provide you with a link.

Thx for looking into this!

syslog server(port:514) started on Thu Jan 05 23:26:03 2012
SDP RTPMAP 101 --> 142
[0:5060]<<10.0.0.201:6060
INVITE sip:900@10.0.0.25:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.0.201:6060;branch=z9hG4bK-fdfa792dFrom: "0475038578"

<0475038578>;tag=d64172115f4c6224o2To: <900>Remote-Party-ID: "0475038578"

<0475038578>;screen=yes;party=callingCall-ID: b9911e55-fff502b0@10.0.0.201CSeq: 101 INVITEMax-Forwards: 70Contact:

Cortex <0475038578>Expires: 240User-Agent: Linksys/SPA9000-6.1.5Allow-Events: talk, hold, conferenceContent-Length:

356Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFERSupported: x-sipura, replacesContent-Type: application/sdpv=0o=proxy

685745591 685745591 IN IP4 85.119.188.67s=Asterisk PBX 1.6.1.25c=IN IP4 10.0.0.201t=0 0m=audio 16446 RTP/AVP 0 8 18 3 101a=rtpmap:0

PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-

16a=silenceSupp:off - - - -a=ptime:20a=sendrecv
[0:5060]->10.0.0.201:6060{285)
SIP/2.0 100 TryingTo: <900>From: "0475038578" <0475038578>;tag=d64172115f4c6224o2Call-ID: b9911e55-

fff502b0@10.0.0.201CSeq: 101 INVITEVia: SIP/2.0/UDP 10.0.0.201:6060;branch=z9hG4bK-fdfa792dServer: Cisco/SPA508G-7.4.9cContent-Length: 0
+++++ find idle call i = 0, caid=0 (asked 0)
[0]CC:NewCallState 10/0

[CCTRL]record lcr phone=0475038578 exten=0 type=0
START_RING in CC_refresh
SDP RTPMAP 101 --> 142
[0:0]AUD ALLOC CALL (port=16526)
[0:5060]->10.0.0.201:6060{434)
SIP/2.0 180 RingingTo: <900>;tag=2a8bb6d47ad84920i0From: "0475038578"

<0475038578>;tag=d64172115f4c6224o2Call-ID: b9911e55-fff502b0@10.0.0.201CSeq: 101 INVITEVia: SIP/2.0/UDP

10.0.0.201:6060;branch=z9hG4bK-fdfa792dContact: <900>Server: Cisco/SPA508G-7.4.9cRemote-Party-ID:

<900>;screen=yes;party=calledContent-Length: 0Allow-Events: dialog
[0:5060]<<10.0.0.201:6060
CANCEL sip:900@10.0.0.25:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.0.201:6060;branch=z9hG4bK-fdfa792dFrom: "0475038578"

<0475038578>;tag=d64172115f4c6224o2To: <900>Call-ID: b9911e55-fff502b0@10.0.0.201CSeq: 101 CANCELMax-

Forwards: 70User-Agent: Linksys/SPA9000-6.1.5Allow-Events: talk, hold, conferenceContent-Length: 0
[0:5060]->10.0.0.201:6060{320)
SIP/2.0 487 Request TerminatedTo: <900>;tag=2a8bb6d47ad84920i0From: "0475038578"

<0475038578>;tag=d64172115f4c6224o2Call-ID: b9911e55-fff502b0@10.0.0.201CSeq: 101 INVITEVia: SIP/2.0/UDP

10.0.0.201:6060;branch=z9hG4bK-fdfa792dServer: Cisco/SPA508G-7.4.9cContent-Length: 0
[0:5060]->10.0.0.201:6060{304)
SIP/2.0 200 OKTo: <900>;tag=2a8bb6d47ad84920i0From: "0475038578"

<0475038578>;tag=d64172115f4c6224o2Call-ID: b9911e55-fff502b0@10.0.0.201CSeq: 101 CANCELVia: SIP/2.0/UDP

10.0.0.201:6060;branch=z9hG4bK-fdfa792dServer: Cisco/SPA508G-7.4.9cContent-Length: 0
[0:0]AUD Rel Call
[CCTRL]record lcr phone=0475038578 exten=0
[0]CC:NewCallState 0/10

[0:5060]<<10.0.0.201:6060
ACK sip:900@10.0.0.25:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.0.201:6060;branch=z9hG4bK-fdfa792dFrom: "0475038578"

<0475038578>;tag=d64172115f4c6224o2To: <900>;tag=2a8bb6d47ad84920i0Call-ID: b9911e55-

fff502b0@10.0.0.201CSeq: 101 ACKMax-Forwards: 70Contact: Cortex <0475038578>User-Agent: Linksys/SPA9000-6.1.5Allow-

Events: talk, hold, conferenceContent-Length: 0
DLG Terminated
Sess Terminated

nseto · ‎01-05-2012

It's working properly, the Invite shows a number and no name.

Remote-Party-ID: "0475038578"

asienaert · ‎01-05-2012

This used to work with my previous phones, SPA962.

I never checked the debug logs on these phones but I'm pretty sure that they were also not receiving the name but they used the number to lookup the phone number in the personal directory, similar to what a cell phone does.

Any other suggestions?

asienaert · ‎01-06-2012

I reconnected an old SPA962 and I can confirm that the phone displays the name it finds in the Personal Directory when an incoming number is found in the directory.

I also enabled debugging and here are the results:

SDP RTPMAP 101 --> 136
[0:0]AUD ALLOC CALL (port=16404)
[0:0]RTP Rx Up
[0:5060]->10.0.0.201:6060{437)
[0:5060]->10.0.0.201:6060{437)
SIP/2.0 180 Ringing

To: <900>;tag=25f5e44263a64d92i0

From: "0475038578" <0475038578>;tag=c4a18a304cce4241o2

Call-ID: 2879701c-c3ffb545@10.0.0.201

CSeq: 101 INVITE

Via: SIP/2.0/UDP 10.0.0.201:6060;branch=z9hG4bK-4b2f149e

Contact: <900>

Server: Linksys/SPA962-6.1.5(a)

Remote-Party-ID: <900>;screen=yes;party=called

Content-Length: 0

Allow-Events: dialog

[0:5060]<<10.0.0.201:6060
[0:5060]<<10.0.0.201:6060
CANCEL sip:900@10.0.0.12:5060 SIP/2.0

Via: SIP/2.0/UDP 10.0.0.201:6060;branch=z9hG4bK-4b2f149e

From: "0475038578" <0475038578>;tag=c4a18a304cce4241o2

To: <900>

Call-ID: 2879701c-c3ffb545@10.0.0.201

CSeq: 101 CANCEL

Max-Forwards: 70

User-Agent: Linksys/SPA9000-6.1.5

Allow-Events: talk, hold, conference

Content-Length: 0

nseto · ‎01-06-2012

Hi, you can open a case with the support center, requesting that the 50x work like the 9x2 series, where it should look up for a match of the phone number to a name in the directory. This will allow for your request to get through to dev.

The support center info is located at

http://www.cisco.com/en/US/support/tsd_cisco_small_business_support_center_contacts.html

Thanks.

asienaert · ‎01-09-2012

@nseto

I opened a case with Cisco support as you suggested.

Thx for your input.