CallerID issue with the Agent interface
Posted: Fri Aug 17, 2012 8:36 amHi gurus,
I have a strange issue with caller id.When i directly call from the xlite or audio codes without logging into the Agent screen,the callerid will popup.
Logs from My carrier
Call Made from Audio code dial pad
U 2012/08/17 17:28:40.715640 202.71.158.38:5060 -> 202.71.134.13:5060
INVITE sip:8903447810181172@202.71.134.13;cpd=on SIP/2.0.
Via: SIP/2.0/UDP 202.71.158.38:5060;branch=z9hG4bK5e66e09d;rport.
From: "2413657814" <sip:052633247@202.71.134.13>;tag=as7ec89fc7.
To: <sip:8903447810181172@202.71.134.13;cpd=on>.
Contact: <sip:052633247@202.71.158.38>.
Call-ID: 613d0e8607ca6082592be08d296e1f51@202.71.134.13.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "442413657814" <sip:2413657814@202.71.134.13>;privacy=off;screen=no.
Proxy-Authorization: Digest username="052633247", realm="202.71.134.13", algorithm=MD5, uri="sip:8903447810181172@202.71.134.13;cpd=on", nonce="bd3665e080df93a880773fe125e5d964", response="b8daafa1474264a9d9e0fe25750cd6a9".
Date: Fri, 17 Aug 2012 11:59:44 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 236.
v=0.
o=root 3033 3034 IN IP4 202.71.158.38.
s=session.
c=IN IP4 202.71.158.38.
t=0 0.
m=audio 10126 RTP/AVP 0 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
Calls Made from Xlite Dialpad
U 2012/08/16 17:12:00.684604 202.71.158.38:5060 -> 202.71.134.13:5060
INVITE sip:8903442085264000@202.71.134.13;cpd=on SIP/2.0.
Via: SIP/2.0/UDP 202.71.158.38:5060;branch=z9hG4bK4d6b5bb9;rport.
From: "442413657814" <sip:052633247@202.71.134.13>;tag=as4cea6c90.
To: <sip:8903442085264000@202.71.134.13;cpd=on>.
Contact: <sip:052633247@202.71.158.38>.
Call-ID: 3236df933a70237a1c01221a35192092@202.71.134.13.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "442413657814" <sip:2413657814@202.71.134.13>;privacy=off;screen=no.
Proxy-Authorization: Digest username="052633247", realm="202.71.134.13", algorithm=MD5, uri="sip:8903442085264000@202.71.134.13;cpd=on", nonce="1533054f2e49860ecf5f783737cadbdb", response="24cc16a9f477daff4a928572ecf4ccf0".
Date: Thu, 16 Aug 2012 11:43:06 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 236.
v=0.
o=root 3027 3028 IN IP4 202.71.158.38.
s=session.
c=IN IP4 202.71.158.38.
t=0 0.
m=audio 13016 RTP/AVP 0 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
U 2012/08/16 17:19:30.208567 202.71.158.38:5060 -> 202.71.134.13:5060
INVITE sip:8903441270308000@202.71.134.13;cpd=on SIP/2.0.
Via: SIP/2.0/UDP 202.71.158.38:5060;branch=z9hG4bK77cf8141;rport.
From: "442413657814" <sip:052633247@202.71.134.13>;tag=as6533580c.
To: <sip:8903441270308000@202.71.134.13;cpd=on>.
Contact: <sip:052633247@202.71.158.38>.
Call-ID: 07fe11ca5b9feb9f79d7a7c10f9bc017@202.71.134.13.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "442413657814" <sip:2413657814@202.71.134.13>;privacy=off;screen=no.
Proxy-Authorization: Digest username="052633247", realm="202.71.134.13", algorithm=MD5, uri="sip:8903441270308000@202.71.134.13;cpd=on", nonce="948bec2ef05bffbc667b561b012c3734", response="5078c6d3df59fd3f610c6686eabb50fd".
Date: Thu, 16 Aug 2012 11:50:35 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 236.
v=0.
o=root 3027 3028 IN IP4 202.71.158.38.
s=session.
c=IN IP4 202.71.158.38.
t=0 0.
m=audio 13002 RTP/AVP 0 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
But when i am calling Manually or Dial ratio from Agent screen my callerid is shown in alpha-numeric values.
SIP DEBUG Output
<------------->
[Aug 17 19:04:51] --- (12 headers 11 lines) ---
[Aug 17 19:04:51]
<--- SIP read from 202.71.134.13:5060 --->
SIP/2.0 200 OK
Call-ID: 1edfc3c15b3d11214352deda3eb6997b@202.71.134.13
CSeq: 103 INVITE
Allow: INVITE,BYE,OPTIONS,CANCEL,ACK,REGISTER,NOTIFY,INFO,REFER,SUBSCRIBE,PRACK,UPDATE,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp
Via: SIP/2.0/UDP 202.71.158.38:5060;branch=z9hG4bK5b599fb0;rport
To: <sip:8903441832710536@202.71.134.13;cpd=on>;tag=3554199211-480212
From: "M8171904310000305465" <sip:052633247@202.71.134.13>;tag=as10201eb0
Contact: <sip:8903441832710536@202.71.134.13:5060;cpd=on>
Remote-Party-ID: <sip:8903441832710536@202.71.134.13:5060;cpd=on>;screen=yes;party=calling;privacy=off
Content-Length: 223
v=0
o=net4-sbc1-2 15620 5307 IN IP4 118.67.254.113
s=sip call
c=IN IP4 116.0.70.5
t=0 0
m=audio 14642 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Aug 17 19:04:51] --- (12 headers 11 lines) ---
[Aug 17 19:04:51] Found RTP audio format 0
[Aug 17 19:04:51] Found RTP audio format 101
[Aug 17 19:04:51] Found audio description format PCMU for ID 0
[Aug 17 19:04:51] Found audio description format telephone-event for ID 101
[Aug 17 19:04:51] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Aug 17 19:04:51] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Aug 17 19:04:51] Peer audio RTP is at port 116.0.70.5:14642
[Aug 17 19:04:51] list_route: hop: <sip:8903441832710536@202.71.134.13:5060;cpd=on>
[Aug 17 19:04:51] set_destination: Parsing <sip:8903441832710536@202.71.134.13:5060;cpd=on> for address/port to send to
[Aug 17 19:04:51] set_destination: set destination to 202.71.134.13, port 5060
[Aug 17 19:04:51] Transmitting (NAT) to 202.71.134.13:5060:
ACK sip:8903441832710536@202.71.134.13:5060;cpd=on SIP/2.0
Via: SIP/2.0/UDP 202.71.158.38:5060;branch=z9hG4bK6fb77fa2;rport
From: "M8171904310000305465" <sip:052633247@202.71.134.13>;tag=as10201eb0
To: <sip:8903441832710536@202.71.134.13;cpd=on>;tag=3554199211-480212
Contact: <sip:052633247@202.71.158.38>
Call-ID: 1edfc3c15b3d11214352deda3eb6997b@202.71.134.13
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M8171904310000305465" <sip:442413657814<442413657814>@202.71.134.13>;privacy=off;screen=no
Content-Length: 0
Vicibox Redux | VERSION: 2.6-370a | BUILD: 120529-2112 | Asterisk 1.4
I have a strange issue with caller id.When i directly call from the xlite or audio codes without logging into the Agent screen,the callerid will popup.
Logs from My carrier
Call Made from Audio code dial pad
U 2012/08/17 17:28:40.715640 202.71.158.38:5060 -> 202.71.134.13:5060
INVITE sip:8903447810181172@202.71.134.13;cpd=on SIP/2.0.
Via: SIP/2.0/UDP 202.71.158.38:5060;branch=z9hG4bK5e66e09d;rport.
From: "2413657814" <sip:052633247@202.71.134.13>;tag=as7ec89fc7.
To: <sip:8903447810181172@202.71.134.13;cpd=on>.
Contact: <sip:052633247@202.71.158.38>.
Call-ID: 613d0e8607ca6082592be08d296e1f51@202.71.134.13.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "442413657814" <sip:2413657814@202.71.134.13>;privacy=off;screen=no.
Proxy-Authorization: Digest username="052633247", realm="202.71.134.13", algorithm=MD5, uri="sip:8903447810181172@202.71.134.13;cpd=on", nonce="bd3665e080df93a880773fe125e5d964", response="b8daafa1474264a9d9e0fe25750cd6a9".
Date: Fri, 17 Aug 2012 11:59:44 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 236.
v=0.
o=root 3033 3034 IN IP4 202.71.158.38.
s=session.
c=IN IP4 202.71.158.38.
t=0 0.
m=audio 10126 RTP/AVP 0 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
Calls Made from Xlite Dialpad
U 2012/08/16 17:12:00.684604 202.71.158.38:5060 -> 202.71.134.13:5060
INVITE sip:8903442085264000@202.71.134.13;cpd=on SIP/2.0.
Via: SIP/2.0/UDP 202.71.158.38:5060;branch=z9hG4bK4d6b5bb9;rport.
From: "442413657814" <sip:052633247@202.71.134.13>;tag=as4cea6c90.
To: <sip:8903442085264000@202.71.134.13;cpd=on>.
Contact: <sip:052633247@202.71.158.38>.
Call-ID: 3236df933a70237a1c01221a35192092@202.71.134.13.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "442413657814" <sip:2413657814@202.71.134.13>;privacy=off;screen=no.
Proxy-Authorization: Digest username="052633247", realm="202.71.134.13", algorithm=MD5, uri="sip:8903442085264000@202.71.134.13;cpd=on", nonce="1533054f2e49860ecf5f783737cadbdb", response="24cc16a9f477daff4a928572ecf4ccf0".
Date: Thu, 16 Aug 2012 11:43:06 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 236.
v=0.
o=root 3027 3028 IN IP4 202.71.158.38.
s=session.
c=IN IP4 202.71.158.38.
t=0 0.
m=audio 13016 RTP/AVP 0 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
U 2012/08/16 17:19:30.208567 202.71.158.38:5060 -> 202.71.134.13:5060
INVITE sip:8903441270308000@202.71.134.13;cpd=on SIP/2.0.
Via: SIP/2.0/UDP 202.71.158.38:5060;branch=z9hG4bK77cf8141;rport.
From: "442413657814" <sip:052633247@202.71.134.13>;tag=as6533580c.
To: <sip:8903441270308000@202.71.134.13;cpd=on>.
Contact: <sip:052633247@202.71.158.38>.
Call-ID: 07fe11ca5b9feb9f79d7a7c10f9bc017@202.71.134.13.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "442413657814" <sip:2413657814@202.71.134.13>;privacy=off;screen=no.
Proxy-Authorization: Digest username="052633247", realm="202.71.134.13", algorithm=MD5, uri="sip:8903441270308000@202.71.134.13;cpd=on", nonce="948bec2ef05bffbc667b561b012c3734", response="5078c6d3df59fd3f610c6686eabb50fd".
Date: Thu, 16 Aug 2012 11:50:35 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 236.
v=0.
o=root 3027 3028 IN IP4 202.71.158.38.
s=session.
c=IN IP4 202.71.158.38.
t=0 0.
m=audio 13002 RTP/AVP 0 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
But when i am calling Manually or Dial ratio from Agent screen my callerid is shown in alpha-numeric values.
SIP DEBUG Output
<------------->
[Aug 17 19:04:51] --- (12 headers 11 lines) ---
[Aug 17 19:04:51]
<--- SIP read from 202.71.134.13:5060 --->
SIP/2.0 200 OK
Call-ID: 1edfc3c15b3d11214352deda3eb6997b@202.71.134.13
CSeq: 103 INVITE
Allow: INVITE,BYE,OPTIONS,CANCEL,ACK,REGISTER,NOTIFY,INFO,REFER,SUBSCRIBE,PRACK,UPDATE,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp
Via: SIP/2.0/UDP 202.71.158.38:5060;branch=z9hG4bK5b599fb0;rport
To: <sip:8903441832710536@202.71.134.13;cpd=on>;tag=3554199211-480212
From: "M8171904310000305465" <sip:052633247@202.71.134.13>;tag=as10201eb0
Contact: <sip:8903441832710536@202.71.134.13:5060;cpd=on>
Remote-Party-ID: <sip:8903441832710536@202.71.134.13:5060;cpd=on>;screen=yes;party=calling;privacy=off
Content-Length: 223
v=0
o=net4-sbc1-2 15620 5307 IN IP4 118.67.254.113
s=sip call
c=IN IP4 116.0.70.5
t=0 0
m=audio 14642 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Aug 17 19:04:51] --- (12 headers 11 lines) ---
[Aug 17 19:04:51] Found RTP audio format 0
[Aug 17 19:04:51] Found RTP audio format 101
[Aug 17 19:04:51] Found audio description format PCMU for ID 0
[Aug 17 19:04:51] Found audio description format telephone-event for ID 101
[Aug 17 19:04:51] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Aug 17 19:04:51] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Aug 17 19:04:51] Peer audio RTP is at port 116.0.70.5:14642
[Aug 17 19:04:51] list_route: hop: <sip:8903441832710536@202.71.134.13:5060;cpd=on>
[Aug 17 19:04:51] set_destination: Parsing <sip:8903441832710536@202.71.134.13:5060;cpd=on> for address/port to send to
[Aug 17 19:04:51] set_destination: set destination to 202.71.134.13, port 5060
[Aug 17 19:04:51] Transmitting (NAT) to 202.71.134.13:5060:
ACK sip:8903441832710536@202.71.134.13:5060;cpd=on SIP/2.0
Via: SIP/2.0/UDP 202.71.158.38:5060;branch=z9hG4bK6fb77fa2;rport
From: "M8171904310000305465" <sip:052633247@202.71.134.13>;tag=as10201eb0
To: <sip:8903441832710536@202.71.134.13;cpd=on>;tag=3554199211-480212
Contact: <sip:052633247@202.71.158.38>
Call-ID: 1edfc3c15b3d11214352deda3eb6997b@202.71.134.13
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M8171904310000305465" <sip:442413657814<442413657814>@202.71.134.13>;privacy=off;screen=no
Content-Length: 0
Vicibox Redux | VERSION: 2.6-370a | BUILD: 120529-2112 | Asterisk 1.4
