Unavailable numbers on voip phone -3CX
Hi
We are using voip phone and provider is 3CX . Currently we are receiving a lot of phones call with unavailable number. 3rd party support is saying that everything is ok from their side. When we ask customer if they are ringing from withheld number , most of them say that they are not ringing from withheld number. Please advice what could be wrong
We are using voip phone and provider is 3CX . Currently we are receiving a lot of phones call with unavailable number. 3rd party support is saying that everything is ok from their side. When we ask customer if they are ringing from withheld number , most of them say that they are not ringing from withheld number. Please advice what could be wrong
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1) Current versions of 3CX< you can do the Wireshark capture from the 3CX "Activity Log" screen.
- Click "capture"
- make your test call
- STOP the capture
- a few seconds later you'll get a URL to download the .pcap file. Use Wireshark on any machine to open this to examine.
2) If you are definitely receiving the sender-CLI but it isn't displaying, you will need to check your "Sip Trunk - Inbound Parameters" tab. There are lots of options here and you can easily break what was working so FIRST take careful notes of the settings as they are now. It's probably going to be best if you check with your SIP trunk supplier to see which fields they're sending. You need to match the "CallerNum" parameter to a suitable source field, usually "From:User Part" but it could be "Remote Party ID - Calling Party: User part", for exampe, or P-Asserted-Identity. NOTE that these should be correct, however, if the sip trunk supplier is 3CX "approved" and you used the correct template to set up the trunk.
- Click "capture"
- make your test call
- STOP the capture
- a few seconds later you'll get a URL to download the .pcap file. Use Wireshark on any machine to open this to examine.
2) If you are definitely receiving the sender-CLI but it isn't displaying, you will need to check your "Sip Trunk - Inbound Parameters" tab. There are lots of options here and you can easily break what was working so FIRST take careful notes of the settings as they are now. It's probably going to be best if you check with your SIP trunk supplier to see which fields they're sending. You need to match the "CallerNum" parameter to a suitable source field, usually "From:User Part" but it could be "Remote Party ID - Calling Party: User part", for exampe, or P-Asserted-Identity. NOTE that these should be correct, however, if the sip trunk supplier is 3CX "approved" and you used the correct template to set up the trunk.
i found a great tool to track COIP traffic. lookup sngrep if it is in your distro. This is a very accessible tool to collect data on SIP.
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