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In DID, SPA3102 is not able to pass Caller ID to FreeSWITCH. Please help.

Dear all,

I wish to capture caller ID for inbound calls. I am using Linksys SPA3102 as PSTN gateway. I want to route the call to FreeSWITCH extension. Routing and calling are working perfectly. The problem is on gateway side as explained below.

When a incomng call comes to SPA3102, it sends SIP INVITE to FreeSWITCH (https://www.freeswitch.org/) by which the call is set up. During that SPA3102 captures CID. If I check syslog, it is shown as below.

FXO:Start CNDD
FXO:Digit=8
FXO:Digit=5
FXO:Digit=5
FXO:Digit=8
FXO:Digit=4
FXO:Digit=2
FXO:Digit=0
FXO:Digit=0
FXO:Digit=2
FXO:Digit=4
AUD:Stop PSTN Tone
AUD:Stop PSTN Tone

But, when INVITE command does not hold this CID (8558420024) as follows.

INVITE sip:8041145676@192.168.7.36 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.82:5062;branch=z9hG4bK-180f9baf
From: <sip:192.168.7.36>;tag=8dc3630596c3033fo1
To: <sip:8041145676@192.168.7.36>
Remote-Party-ID: <sip:192.168.7.36>;screen=yes;party=calling
Call-ID: 1cfd3afd-d3b64ba7@192.168.7.82
CSeq: 101 INVITE
Max-Forwards: 70
Contact: <sip:192.168.7.82:5062>
Expires: 240
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 516097 516097 IN IP4 192.168.7.82
s=-
c=IN IP4 192.168.7.82
t=0 0
m=audio 16454 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

Therefore, FreeSWITCH could not identify caller (displays 'nobody'). One line of debug log from FreeSWITCH is given below.

946234d8-b034-11e3-b63b-37f3148ad12d 2014-03-20 19:10:02.549490 [NOTICE] switch_channel.c:1053 New Channel sofia/internal/nobody@192.168.7.36 [946234d8-b034-11e3-b63b-37f3148ad12d]

Selective portion of log files from SPA3102 and FreeSWITCH and SPA3102 configuration are given as an attachments.

Could you please help on the matter? I need the CID at FreeSWITCH which SPA3102 has to pass.

Many thanks

Debkumar

 

SPA3102 Log
------------------------------------------------------------------
FXO:Start CNDD
FXO:Digit=8
FXO:Digit=5
FXO:Digit=5
FXO:Digit=8
FXO:Digit=4
FXO:Digit=2
FXO:Digit=0
FXO:Digit=0
FXO:Digit=2
FXO:Digit=4
AUD:Stop PSTN Tone
AUD:Stop PSTN Tone
Calling:8041145676@192.168.7.36:0
[1:0]AUD ALLOC CALL (port=16454)
[1:0]RTP Rx Up
[1]->192.168.7.36:5060(1004)
[1]->192.168.7.36:5060(1004)
INVITE sip:8041145676@192.168.7.36 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.82:5062;branch=z9hG4bK-180f9baf
From: <sip:192.168.7.36>;tag=8dc3630596c3033fo1
To: <sip:8041145676@192.168.7.36>
Remote-Party-ID: <sip:192.168.7.36>;screen=yes;party=calling
Call-ID: 1cfd3afd-d3b64ba7@192.168.7.82
CSeq: 101 INVITE
Max-Forwards: 70
Contact: <sip:192.168.7.82:5062>
Expires: 240
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 516097 516097 IN IP4 192.168.7.82
s=-
c=IN IP4 192.168.7.82
t=0 0
m=audio 16454 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv


[1]<<192.168.7.36:5060(324)
[1]<<192.168.7.36:5060(324)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.82:5062;branch=z9hG4bK-180f9baf
From: <sip:192.168.7.36>;tag=8dc3630596c3033fo1
To: <sip:8041145676@192.168.7.36>
Call-ID: 1cfd3afd-d3b64ba7@192.168.7.82
CSeq: 101 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140319T030354Z~1723c12ede~32bit
Content-Length: 0

 

[1]<<192.168.7.36:5060(1117)
[1]<<192.168.7.36:5060(1117)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.82:5062;branch=z9hG4bK-180f9baf
From: <sip:192.168.7.36>;tag=8dc3630596c3033fo1
To: <sip:8041145676@192.168.7.36>;tag=m5geDvp9S51mj
Call-ID: 1cfd3afd-d3b64ba7@192.168.7.82
CSeq: 101 INVITE
Contact: <sip:8041145676@192.168.7.36:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140319T030354Z~1723c12ede~32bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 220
Remote-Party-ID: "8041145676" <sip:8041145676@192.168.7.36>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1395291176 1395291177 IN IP4 192.168.7.36
s=FreeSWITCH
c=IN IP4 192.168.7.36
t=0 0
m=audio 31598 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


[1]->192.168.7.36:5060(390)
[1]->192.168.7.36:5060(390)
ACK sip:8041145676@192.168.7.36:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.7.82:5062;branch=z9hG4bK-aa17129f
From: <sip:192.168.7.36>;tag=8dc3630596c3033fo1
To: <sip:8041145676@192.168.7.36>;tag=m5geDvp9S51mj
Call-ID: 1cfd3afd-d3b64ba7@192.168.7.82
CSeq: 101 ACK
Max-Forwards: 70
Contact: <sip:192.168.7.82:5062>
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0

 

[1:0]ENC INIT 0
[1:0]RTP Tx Up (pt=0->c0a80724:31598)
[1:0]RTCP Tx Up
CC:Remote Resume
AUD:Stop PSTN Tone
CC:Connected
AUD:Stop PSTN Tone
FXO:Off Hook
FXO:Stop CNDD
[1:0]RTP Rx 1st PKT @16454(2)
[1:0]DEC INIT 0
[1]<<192.168.7.36:5060(632)
[1]<<192.168.7.36:5060(632)
BYE sip:192.168.7.82:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.36;rport;branch=z9hG4bKH2tm6Fe9ZrtjK
Max-Forwards: 70
From: <sip:8041145676@192.168.7.36>;tag=m5geDvp9S51mj
To: <sip:192.168.7.36>;tag=8dc3630596c3033fo1
Call-ID: 1cfd3afd-d3b64ba7@192.168.7.82
CSeq: 57325352 BYE
Contact: <sip:8041145676@192.168.7.36:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140319T030354Z~1723c12ede~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0

 

[1]->192.168.7.36:5060(294)
[1]->192.168.7.36:5060(294)
SIP/2.0 200 OK
To: <sip:192.168.7.36>;tag=8dc3630596c3033fo1
From: <sip:8041145676@192.168.7.36>;tag=m5geDvp9S51mj
Call-ID: 1cfd3afd-d3b64ba7@192.168.7.82
CSeq: 57325352 BYE
Via: SIP/2.0/UDP 192.168.7.36;branch=z9hG4bKH2tm6Fe9ZrtjK
Server: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0

 

CC:Ended
AUD:Stop PSTN Tone
[0]FM Alert Stop RxTx (c=0024fe9c;a=0)
[1:0]AUD Rel Call
AUD:Stop PSTN Tone
FXO:On Hook
AUD:Stop PSTN Tone
FXO:Stop CNDD
AUD:Stop PSTN Tone
DLG Terminated 29bb68
Sess Terminated
CC:Clean Up
--- OBJ POOL STAT ---
OP:RTPRXB =  96 ( 96  192)   OP:RTPREB =  40 ( 40   48)
OP:RTPTXB =  64 ( 64  108)   OP:TIMEOU = 112 (120   52)
OP:SIPCOR =   0 (  1   28)   OP:SIPCTS =  32 ( 32 1140)
OP:SIPSTS =  32 ( 32 6620)   OP:SIPAUS =   8 (  8  588)
OP:SIPDLG =  10 ( 10  148)   OP:SIPSES =  12 ( 12 9512)
OP:SIPREG =   4 (  4  296)   OP:SIPLIN =   0 (  2  140)
OP:SUBDLG =   2 (  2 6436)   OP:STUNTS =  16 ( 16   68)
OP:XMNODE = 1024 (1024  112)   

 


FreeSWITCH Log
-----------------------------------------------------------------------

 

946234d8-b034-11e3-b63b-37f3148ad12d 2014-03-20 19:10:02.549490 [NOTICE] switch_channel.c:1053 New Channel sofia/internal/nobody@192.168.7.36 [946234d8-b034-11e3-b63b-37f3148ad12d]
946234d8-b034-11e3-b63b-37f3148ad12d 2014-03-20 19:10:02.549490 [DEBUG] switch_core_session.c:1050 Send signal sofia/internal/nobody@192.168.7.36 [BREAK]
946234d8-b034-11e3-b63b-37f3148ad12d 2014-03-20 19:10:02.549490 [DEBUG] switch_core_session.c:1050 Send signal sofia/internal/nobody@192.168.7.36 [BREAK]
946234d8-b034-11e3-b63b-37f3148ad12d 2014-03-20 19:10:02.549490 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/nobody@192.168.7.36) Running State Change CS_NEW
946234d8-b034-11e3-b63b-37f3148ad12d 2014-03-20 19:10:02.549490 [DEBUG] sofia.c:8293 sofia/internal/nobody@192.168.7.36 receiving invite from 192.168.7.82:5062 version: 1.5.12b git 1723c12 2014-03-19 03:03:54Z 32bit
2014-03-20 19:10:02.549490 [DEBUG] sofia.c:8405 IP 192.168.7.82 Approved by acl "domains[]". Access Granted.
946234d8-b034-11e3-b63b-37f3148ad12d 2014-03-20 19:10:02.549490 [DEBUG] sofia.c:6163 Channel sofia/internal/nobody@192.168.7.36 entering state [received][100]
946234d8-b034-11e3-b63b-37f3148ad12d 2014-03-20 19:10:02.549490 [DEBUG] sofia.c:6173 Remote SDP:
946234d8-b034-11e3-b63b-37f3148ad12d v=0
946234d8-b034-11e3-b63b-37f3148ad12d o=- 494940 494940 IN IP4 192.168.7.82
946234d8-b034-11e3-b63b-37f3148ad12d s=-
946234d8-b034-11e3-b63b-37f3148ad12d c=IN IP4 192.168.7.82
946234d8-b034-11e3-b63b-37f3148ad12d t=0 0
946234d8-b034-11e3-b63b-37f3148ad12d m=audio 16452 RTP/AVP 0 2 4 8 18 96 97 98 100 101
946234d8-b034-11e3-b63b-37f3148ad12d a=rtpmap:0 PCMU/8000
946234d8-b034-11e3-b63b-37f3148ad12d a=rtpmap:2 G726-32/8000
946234d8-b034-11e3-b63b-37f3148ad12d a=rtpmap:4 G723/8000
946234d8-b034-11e3-b63b-37f3148ad12d a=rtpmap:8 PCMA/8000
946234d8-b034-11e3-b63b-37f3148ad12d a=rtpmap:18 G729a/8000
946234d8-b034-11e3-b63b-37f3148ad12d a=rtpmap:96 G726-40/8000
946234d8-b034-11e3-b63b-37f3148ad12d a=rtpmap:97 G726-24/8000
946234d8-b034-11e3-b63b-37f3148ad12d a=rtpmap:98 G726-16/8000
946234d8-b034-11e3-b63b-37f3148ad12d a=rtpmap:100 NSE/8000
946234d8-b034-11e3-b63b-37f3148ad12d a=fmtp:100 192-193
946234d8-b034-11e3-b63b-37f3148ad12d a=rtpmap:101 telephone-event/8000
946234d8-b034-11e3-b63b-37f3148ad12d a=fmtp:101 0-15
946234d8-b034-11e3-b63b-37f3148ad12d a=ptime:20
946234d8-b034-11e3-b63b-37f3148ad12d
946234d8-b034-11e3-b63b-37f3148ad12d 2014-03-20 19:10:02.549490 [DEBUG] sofia.c:6428 (sofia/internal/nobody@192.168.7.36) State Change CS_NEW -> CS_INIT
........
........
127cab5a-b035-11e3-b646-37f3148ad12d EXECUTE sofia/internal/nobody@192.168.7.36 info()
127cab5a-b035-11e3-b646-37f3148ad12d 2014-03-20 19:10:32.289490 [INFO] mod_dptools.c:1621 CHANNEL_DATA:
127cab5a-b035-11e3-b646-37f3148ad12d Channel-State: [CS_EXECUTE]
127cab5a-b035-11e3-b646-37f3148ad12d Channel-Call-State: [ACTIVE]
127cab5a-b035-11e3-b646-37f3148ad12d Channel-State-Number: [4]
127cab5a-b035-11e3-b646-37f3148ad12d Channel-Name: [sofia/internal/nobody@192.168.7.36]
127cab5a-b035-11e3-b646-37f3148ad12d Unique-ID: [127cab5a-b035-11e3-b646-37f3148ad12d]
127cab5a-b035-11e3-b646-37f3148ad12d Call-Direction: [inbound]
127cab5a-b035-11e3-b646-37f3148ad12d Caller-Caller-ID-Name: [unknown]
127cab5a-b035-11e3-b646-37f3148ad12d Caller-Orig-Caller-ID-Name: [unknown]
127cab5a-b035-11e3-b646-37f3148ad12d Caller-Network-Addr: [192.168.7.82]
127cab5a-b035-11e3-b646-37f3148ad12d Caller-Destination-Number: [8041145676]
127cab5a-b035-11e3-b646-37f3148ad12d Caller-Unique-ID: [127cab5a-b035-11e3-b646-37f3148ad12d]

Everyone's tags (2)
1 REPLY
New Member

 I think when the call comes

 I think when the call comes on your TGW, it doesn't forward caller-id to the end points so you need to check your config to allow.

 In general the caller-id is not forwarded to end points/ ip phones and been striped off/stored at TGW.

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