Hi alls,
I register a VoIP gateway to a SIP trunk of Asterisk server.
If I place a call from outbound (out of the VoIP gateway, for example, in PSTN), could I keep the caller ID of outbound caller to display on IP phones registering to Asterisk?
I meet a problem that when I place a call from outbound, the caller ID display on IP phones is the name of the SIP trunk on Asterisk.
How can I correct this problem?
Please give me your comments.
Thanks for all your replys.
That depends on whether the provider supports Remote-Party-ID headers. If they do, enable trust for them.
What device I need to set?
VoIP gateway? or Asterisk?
Both probably. Depends on whether the upstream device is already suitably configured.
However, if you control the upstream device, why configure it to require Asterisk to register?
[quote=“david55”]Both probably. Depends on whether the upstream device is already suitably configured.
However, if you control the upstream device, why configure it to require Asterisk to register?[/quote]
I can control both device.
The Voice gateway need to register to two Asterisk server for redundant. I can not register the Voice gateway as SIP account to Asterisk. Because, I think all the caller id from Voice gateway will be the name of SIP account on all IP phones.
So I think I can register the Voice gateway to a SIP trunk of Asterisk server. But I don’t know could I keep the originated caller ID from outbound so.
Configure the gateway to generate Remote-Party-ID headers; configure Asterisk to accept them. If the gateway can’t do so, change your design.
For Asterisk this is:
[quote=“david55”]Configure the gateway to generate Remote-Party-ID headers; configure Asterisk to accept them. If the gateway can’t do so, change your design.
For Asterisk this is:
Is it in sip.conf file?
Please read the documentation, or just try it and check the log for errors.
Really? Your response was “read the documentation” to the questions “which conf file?”
For what it’s worth, trustrpid is not in the index of “Asterisk: The Definitive Guide”.
Non-answers like yours are why I hate reading internet forums.
The documentation is the sample configuration files. The appendices of the Definitive Guide are basically that reformatted, plus the output from core show application xxxx, etc. Also, if the index is anything like that in TFOT, you are better off doing free text searches on the PDF or HTML.
Generally, for free support, you can expect to get pointers to the information, but not worked examples. We are trying to make you do the leg work.
I am using FreePBX to configure the SIP trunks. What worked for me was to add
…to the outbound side, under “Outgoing Settings”, and
…to the inbound side, under “Incoming Settings”.
HTH YMMV.