I have an HT813 connected to a PSTN line. When an inbound call comes in, whatever i change at settings, i can only see the SIP User ID as the caller id and not the real Caller ID.
The HT813 is registered to a UCM6301 as peer Sip Trunk. There is an inbound route to call an extension by an incoming call. SIP User ID and Authenticate ID is 1111, and these numbers are shown on the end device.
Is there a workaround to fix this issue? The location is in Austria.
Thanks in advance.
you can’t connect the PSTN line directlyhttps://helpdesk.grandstream.com/ to UCM6301’s FXO?
The HT813 and the UCM are in different cities. The UCM has its own PSTN line connected, which works perfectly.
if you don’t give details, the reader certainly won’t know,
first transform the PSTN line into a VoIP trunk and register it on UCM.
As i wrote in my initial post, the HT813 is already registered as a peer sip trunk. Inbound calls are accepted and forwarded to an extension. The only problem is the caller id. Whenever somebody is calling me, the only thing i can see is that 1111 is calling, which number was defined at the HT813 FXO settings page under SIP User ID, Authenticate ID and Name.
sorry, i cant understand your last post. I try to explain the whole setup:
Place A.:
Place B.
At the UCM, i registered the analog trunk (its own FXO port) and a VoIP Trunk (the HT813 as peer SIP Trunk).
Both trunks are functioning great, accepting the incoming calls. For both trunks there is an inbound route registered. So calls are coming in, there is a connection, even the FXS port is working great at place B on the HT813. Only problem is the Caller ID: Incoming call from the PSTN line from Provider “B” is seen as “1111”, which is the SIP User ID given on the FXO setting page at HT813.
Hope this clarifies the situation. Thank you!
with OpenVPN I give up, in various situations I found myself with OpenVPN made by others, removed and put IPsec VPN, everything started to work regularly,
regarding the pstn, as already mentioned, I would switch to trnk voip and register directly to UCM, solve every problem without stomach aches.
I agree with @damiano70 first recommendation, to change the PSTN at Site B from an analog physical phone line to be a VOIP. Perhaps the telephone provider at Site B can do that change. If you have to move the telephone number to a different service provider, in the USA we call that Porting the Number. The VOIP can then be used directly by the UCM6301 at Site A, which eliminates the HT813 and OpenVPN.
You may have a reason to stay with the analog telephone service at Site B.
Have you confirmed that at Site B you are receiving valid Caller ID data? That can be done easily by connecting that telephone line to a analog telephone that has a display. Once this is confirmed you can proceed with testing to find out where the Caller ID info gets changed to the SIP User ID.
Rick
At the UCM you need to do a network capture of a call coming in from the HT. Using Wireshark you can examine the capture and see which fields contain the 1111 and which have the CID of the incoming call.
And of course, there is always the manual for the device that shows how the CID can be transported to the PBX which can be found on page 65 -
According to customer’s choice CID information will be transferred from PSTN network
to VoIP network using following rules:
• Relay via SIP From - PSTN CID is in the SIP From field
• Relay via P-Asserted-Identity - SIP From field uses the pre-configured
account user Id. PSTN CID is in the P-Asserted-Identity field
• Relay via P-Preferred-Identity - PSTN CID is in the P-Preferred-Identity field
• Send anonymous - SIP From field uses “anonymous”. PSTN CID is put in the
P-Asserted-Identity field
• Disable - PSTN CID will not be sent. SIP From field uses the pre-configured
account user ID