Hello,
I have a Dialplan that does the following.
[from-internal]
include => out-route
[out-route]
exten => _0ZXXXXX.,1,Gosub(dial-out,s,1(${EXTEN})
same => n,Hangup()
[dial-out]
exten => s,1,set(CALLERID(num)=${ARG1})
same => n,set(Set(CONNECTEDLINE(name,i)=CID:${ARG1})
same => n,Dial(PJSIP/trunkoutbound/${ARG1})
same => n,return()
When the user dials the number the CallerID in the phone display is correct, as soon as the Dial continues i receive a SIP 183 Progress message the CallerID(num) changes, this because then Pai header has the wrong part and contains the āsā extension.
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.51.230:5060;rport=55409;received=95.97.59.139;branch=z9hG4bK2310283810
Call-ID: 1_1037424360@192.168.51.230
From: "MT-301" <sip:10001-001@pbx.host.eu>;tag=3055177252
To: <sip:0123456789@pbx.host.eu>;tag=mRj05FfphYf99V4CRK0GCnJBATATznqX
CSeq: 2 INVITE
Server: PBX host.EU
Contact: <sip:87.1.2.3:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
P-Asserted-Identity: "CID:0123456789" <sip:s@pbx.host.eu>
Content-Type: application/sdp
Content-Length: 251
Even when is set the CallerID or CONNECTEDLINE before the Dial the part still holds the dialplan exten āsā , Even a DumpChan gives back the CallerID and ConnectedLineID as set bij the diaplan.
Has anyone have a solution for this, or knows if there is a bug report?
Devices are PJSIP and have context=from-internal
using the PJSIP channel driver.
Asterisk version = 13.17.2
Thanks