Hi 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
CallerID issue with the Agent interface
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CallerID issue with the Agent interface
by sunnymannava » Fri Aug 17, 2012 8:36 am
Vicibox Standard 6.0 | 2 SQL Server. 12 Cores. 16 gig Ram. 8 146 gig HD Raid 10 | 3 Web Servers. 12 Cores. 16 gig Ram. 2 73 gig HD Raid 0 | 3 Telephony Servers VERSION: 2.10-452a | VERSION: 2.12-491a-BUILD: 150610-0934
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- Posts: 9
- Joined: Fri May 21, 2010 9:46 am
Re: CallerID issue with the Agent interface
by Acidshock » Tue Aug 21, 2012 5:21 pm
Whats the caller-ID in the campaign screen? Manual dial will use the caller-ID located in the current campaign. That callerid in the campaign settings cannot contain anything but numeric data.
VERSION: 2.14-698a | BUILD: 190207-2301 | Asterisk:13.24.1-vici | Vicibox 8.1.2
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- Posts: 430
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Re: CallerID issue with the Agent interface
by sunnymannava » Wed Aug 22, 2012 2:38 am
The caller id in campaing settings is 442413657814.but it is not passing through.
Vicibox Standard 6.0 | 2 SQL Server. 12 Cores. 16 gig Ram. 8 146 gig HD Raid 10 | 3 Web Servers. 12 Cores. 16 gig Ram. 2 73 gig HD Raid 0 | 3 Telephony Servers VERSION: 2.10-452a | VERSION: 2.12-491a-BUILD: 150610-0934
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- Posts: 9
- Joined: Fri May 21, 2010 9:46 am
Re: CallerID issue with the Agent interface
by williamconley » Wed Apr 10, 2013 12:35 pm
sunnymannava wrote:The caller id in campaing settings is 442413657814.but it is not passing through.
given the odds, your carrier is setting your callerid for you instead of passing it through. ask your carrier.
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Newest Product: Vicidial Agent Only Beep - Beta
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