... but it's not in the siptrace at all...

Can we get more lines of the fs_cli log? From the setting of CID to the actual log?

-Avi

On Tue, Jul 2, 2013 at 3:10 AM, Moishe Grunstein <max@nysolutions.com> wrote:

What type of caller ID format is your ITSP expecting? Who is the ITSP?

http://wiki.freeswitch.org/wiki/Channel_Variables#Caller_ID_Related

 

 

 

Thanks,

 

Moishe Grunstein

Tornado Computer Systems, Inc.

212.400.7650 888.IPPBX.US
Service Request Email: support@nysolutions.com

Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified

cid:image001.jpg@01C72F94.9EE45D60

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

 

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Lloyd Aloysius
Sent: Monday, July 01, 2013 7:39 PM
To: FreeSWITCH Users Help
Subject: spam>spam>[Freeswitch-users] effective_caller_id_name &effective_caller_id_number - not working

 

Hi All,

 

When sending calls out from gateway the caller id name and number not working. Please find the freeswitch cli log and sip invite trace. Any help is appreciated.

 

freeswitch cli

 

EXECUTE sofia/sipinterface_1/barbara@alcan.mydomain.ca set(effective_caller_id_name=Alcan Cable)

2013-07-01 19:23:21.676990 [DEBUG] mod_dptools.c:1373 sofia/sipinterface_1/barbara@alcan.mydomain.ca SET [effective_caller_id_name]=[Alcan Cable]

EXECUTE sofia/sipinterface_1/barbara@alcan.mydomain.ca set(effective_caller_id_number=9052066900)

2013-07-01 19:23:21.676990 [DEBUG] mod_dptools.c:1373 sofia/sipinterface_1/barbara@alcan.mydomain.ca SET [effective_caller_id_number]=[9052066900]

 

SIP Trace

 

 

 

Via: SIP/2.0/UDP 192.168.240.105:5060;branch=z9hG4bK17bfd0ad4

 

Max-Forwards: 70

 

Content-Length: 285

 

 

From: Barbara Ladner <sip:barbara@alcan.mydomain.ca>;tag=02f4abe9bef3a7f

 

 

CSeq: 1894816474 INVITE

 

Supported: timer

 

Allow-Events: talk,hold,conference

 

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

 

Content-Type: application/sdp

 

 

Supported: replaces

 

User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

 

 

 

v=0

 

o=MxSIP 0 575681134 IN IP4 192.168.240.105

 

s=SIP Call

 

c=IN IP4 192.168.240.105

 

t=0 0

 

m=audio 3000 RTP/AVP 0 8 18 101

 

a=rtpmap:0 PCMU/8000

 

a=rtpmap:8 PCMA/8000

 

a=rtpmap:18 G729/8000

 

a=rtpmap:101 telephone-event/8000

 

a=fmtp:101 0-15

 

a=ptime:20

 

a=silenceSupp:off - - - -

 

 

SIP/2.0 100 Trying

 

Via: SIP/2.0/UDP 192.168.240.105:5060;branch=z9hG4bK17bfd0ad4;received=99.247.64.220;rport=16511

 

From: Barbara Ladner <sip:barbara@alcan.mydomain.ca>;tag=02f4abe9bef3a7f

 

 

 

CSeq: 1894816474 INVITE

 

User-Agent: FreeSWITCH-mod_sofia/1.5.3b+git~20130628T160527Z~a52a604fbb

 

Content-Length: 0

 

 

Thanks

Lloyd


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