Applicability Firmware version: Any Model: S-Series Problem Description User was not able to make outbound or inbound calls through SIP trunk. Analysis In PCAP file for a test outgoing call, pbx has sent INVITE to SIP provider but it didn’t reply any message to PBX, in this case it’s better to check with the provider. […]
Applicability Firmware version: Any Model: S-Series, U-Series Problem Description When receiving the incoming call through the SIP trunk, S-series PBX reply ‘421 Extension Required’, then the incoming call fails. Analysis The PCAP log shows S-Series PBX refused the call, then required ‘Timer’, which means S-series PBX requires the SIP provider sends ‘session timer’ at the […]
Applicability Firmware version: Any Model: S-Series Problem DescriptionCreate one SIP peer trunk which applied from carrier, the trunk status shows up. But incoming call can not land to PBX, outgoing call from PBX to external number works fine. AnalysisAnalysis on the PCAP captured on PBX side.As the following figure indicated, PBX respond “Destination Unreachable” based […]
Applicability Firmware version: Any Model: S-Series, U-Series Problem Description User failed to make the internal calls. Analysis The PCAP log shows the RTP is encrypt in SDP that sent to the phone. It’s highly possible that the SRTP is not enabled on the phone when you registered the extension. Solution You can choose either solution: […]
Applicability Firmware version: Any Model: TA FXS Problem Description Client reported that he could not make calls from TA3200 FXS port. Receiving calls was working fine. TA3200 was provisioned by S100. Analysis SIP log on S100 shows error: PJSIP syntax error exception when parsing ‘ ‘ headers on line 4 col 36; The problem comes […]
Applicability Firmware version: Any Model: U-Series & Gateways Problem Description The problem could be like: The gateway is interconnected with Cisco CUCM by SIP trunk. The one-way audio issue happens when the call is retrieved from hold by CUCM extension. One-way audio after call established. In the case of One-way Audio after Hold Retrieve from […]
Users might meet the SIP Call issue including: One way audio No audio in both way Call hangs up at 30 seconds Mostly it is relevant to the router, firewall, or VPN gateway settings. We often suggest client disable SIP ALG, SIP forwarding or something like that. Specifically the one-way audio or no audio issue. […]
Overview Problem Symptom Problem Analysis Why it is 30 seconds? How can we check and analyze the issue? 1 – General Troubleshooting Advice 2 – Problem Information Collection 3 – Log Capture 4 – VoIP Call Analysis in Wireshark. Problem Resolution Resolution How do the PBX NAT settings work? The other Typical Case Problem Description […]
1. Caused by diversion header Set Diversion to None in the setting: Trunk > Advanced > Outbound Parameters, Diversion 2. Incorrect Caller ID number in sip message Set the trunk number in the Caller ID Number field or PBX will send the original extension out to the provider. SIP Trunk> Basic> Caller ID Number 3. […]
Network address translation (NAT) is a method of remapping one IP address space into another by modifying network address information in the IP header of packets while they are in transit across a traffic routing device. The technique was originally used as a shortcut to avoid the need to readdress every host when a network […]