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
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
INVITE sip:4164750401@alcan.mydomain.ca SIP/2.0
Via: SIP/2.0/UDP 192.168.240.105:5060;branch=z9hG4bK17bfd0ad4
Max-Forwards: 70
Content-Length: 285
To: 4164750401 <sip:4164750401@alcan.mydomain.ca>
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
Contact: Barbara Ladner <sip:barbara@192.168.240.105:5060;transport=udp>
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
To: 4164750401 <sip:4164750401@alcan.mydomain.ca>
CSeq: 1894816474 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.3b+git~20130628T160527Z~a52a604fbb
Content-Length: 0
Thanks
Lloyd