We have had less problems with our calls being blocked dialing out since adding sendrpid = yes however it is still not correct. Our Caller ID now shows up as 10 digits instead of 11 even though I have 11 digits entered in the Campaign Caller ID Field.
[Sep 13 08:21:25] -- Executing [51801xxxxxxx@default:1] AGI("Local/8600064@default-0000006f;1", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 13 08:21:25] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TEST1))
[Sep 13 08:21:25] -- <Local/8600064@default-0000006f;1>AGI Script
agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 13 08:21:25] -- Executing [51801xxxxxxx@default:2] Dial("Local/8600064@default-0000006f;1", "SIP/bandwidth/+1801xxxxxxx,,tTo") in new stack
[Sep 13 08:21:25] == Using SIP RTP CoS mark 5
[Sep 13 08:21:25] Audio is at 17122
[Sep 13 08:21:25] Adding codec ulaw to SDP
[Sep 13 08:21:25] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 13 08:21:25] Reliably Transmitting (no NAT) to xx.xx.xx.xx:5060:
[Sep 13 08:21:25] INVITE sip:+1801xxxxxxx@xx.xx.xx.xx:5060 SIP/2.0
[Sep 13 08:21:25] Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK2ad6c1bf
[Sep 13 08:21:25] Max-Forwards: 70
[Sep 13 08:21:25] From: "M9130821250002084220" <sip:1800xxxxxxx@xx.xx.xx.xx>;tag=as35aef776
[Sep 13 08:21:25] To: <sip:+1801xxxxxxx@xx.xx.xx.xx:5060>
[Sep 13 08:21:25] Contact: <sip:1800xxxxxxx@xx.xx.xx.xx:5060>
[Sep 13 08:21:25] Call-ID:
61d001fa1ea0253a77aa4b7578046f17@xx.xx.xx.xx:5060
[Sep 13 08:21:25] CSeq: 102 INVITE
[Sep 13 08:21:25] User-Agent: Asterisk PBX 13.21.1-vici
[Sep 13 08:21:25] Date: Fri, 13 Sep 2019 14:21:25 GMT
[Sep 13 08:21:25] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep 13 08:21:25] Supported: replaces, timer
[Sep 13 08:21:25] Remote-Party-ID: "M9130821250002084220" <sip:1800xxxxxxx@xx.xx.xx.xx>;party=calling;privacy=off;screen=no
[Sep 13 08:21:25] Content-Type: application/sdp
[Sep 13 08:21:25] Content-Length: 259
[Sep 13 08:21:25]
[Sep 13 08:21:25] v=0
[Sep 13 08:21:25] o=root 1486936172 1486936172 IN IP4 xx.xx.xx.xx
[Sep 13 08:21:25] s=Asterisk PBX 13.21.1-vici
[Sep 13 08:21:25] c=IN IP4 xx.xx.xx.xx
[Sep 13 08:21:25] t=0 0
[Sep 13 08:21:25] m=audio 17122 RTP/AVP 0 101
[Sep 13 08:21:25] a=rtpmap:0 PCMU/8000
[Sep 13 08:21:25] a=rtpmap:101 telephone-event/8000
[Sep 13 08:21:25] a=fmtp:101 0-16
[Sep 13 08:21:25] a=ptime:20
[Sep 13 08:21:25] a=maxptime:150
[Sep 13 08:21:25] a=sendrecv
[Sep 13 08:21:25]
[Sep 13 08:21:25] ---
[Sep 13 08:21:25] -- Called SIP/bandwidth/+1801xxxxxxx
[Sep 13 08:21:25]
[Sep 13 08:21:25] <--- SIP read from UDP:xx.xx.xx.xx:5060 --->
[Sep 13 08:21:25] SIP/2.0 100 Giving a try
[Sep 13 08:21:25] Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK2ad6c1bf
[Sep 13 08:21:25] From: "M9130821250002084220" <sip:1800xxxxxxx@xx.xx.xx.xx>;tag=as35aef776
[Sep 13 08:21:25] To: <sip:+1801xxxxxxx@xx.xx.xx.xx:5060>
[Sep 13 08:21:25] Call-ID:
61d001fa1ea0253a77aa4b7578xxxxx@xx.xx.xx.xx:5060
[Sep 13 08:21:25] CSeq: 102 INVITE
[Sep 13 08:21:25] Server: Bandwidth.com TRM (bw7.gold.13)
[Sep 13 08:21:25] Content-Length: 0