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Outbound caller ID issue with fromuser in sip peers
Posted:
Fri Dec 06, 2019 9:56 am
by Jitin
Hello Sir
Asterisk version : 11.22.0
VERSION: 2.14-694a
BUILD: 181005-1738
OS : centos 6.10
It is a scratch install.
I have issue with caller ID.
the VoIP carrier that I am using is from magnus biling and I need to set "fromuser" in sip peers.
With fromuser, the caller ID goes what is set in fromuser. Is there anyway to bypass the caller ID?
I want to send caller ID which is set in campaign in vici as we are also using ac-cid.
Re: Outbound caller ID issue with fromuser in sip peers
Posted:
Fri Dec 06, 2019 6:35 pm
by ambiorixg12
If they set the caller id based on the Remote-Party-I header set sendrpid = yes on your carrier configuration
Re: Outbound caller ID issue with fromuser in sip peers
Posted:
Mon Dec 09, 2019 1:48 am
by Jitin
That didn't make any difference. Caller ID id still going from fromuser. Without fromuser , sip account is not working to send calls. SO need an alternative for the same.
Re: Outbound caller ID issue with fromuser in sip peers
Posted:
Mon Dec 09, 2019 6:24 am
by Jitin
This was resolved
had to add trustpid at carrier server.
and sendrpid=pai at dialer end in carrier settings.
thanxs for the support.
Re: Outbound caller ID issue with fromuser in sip peers
Posted:
Thu Mar 05, 2020 5:59 pm
by williamconley
Excellent postback.
Re: Outbound caller ID issue with fromuser in sip peers
Posted:
Thu Nov 18, 2021 8:58 am
by carrmic
I found this trouble too when my SIP trunk with Twilio needed “fromuser” set. The answer was to put “sendrpid=pai” in the Trunk Peer Details. I encountered it again when I set up a new PJSIP Trunk. For this type there are switches in the advanced settings for the trunk. I needed to set “Trust RPID/PAI” to “Yes” and also set “Send RPID/PAI” to “Send P-Asserted-Identity header”. I was then able to pass the caller ID from the extension’s setting. CID set at Trunk or Outbound Route will also pass through.
I’ll add that I’m not sure why I didn’t need “trustrpid=yes” in the Peer Details for the standard SIP trunk, but did need to set that switch in the PJSIP trunk.