We have the phone system divert to a mobile phone after hours. This was working fine until the audio stopped working. When we ring that particular mobile number from the phone system, it doesn’t ring, but it does answer and then there is no audio either way. We can ring any other mobile number fine. I checked the logs and saw no issues there. I then did a tcpdump and included RTP. I then played the stream back in Wireshark and I can hear it ring and the person on the other end answering.
Can someone please shed some light on this because I am stumped.
Kind Regards,
Pieter
Did the capture show the RTP from the trunk being sent out to the extension (with correct IP address and port)? If transcoding is involved, does the outbound audio sound ok?
How about the other direction (is the RTP sent to the trunk have the correct destination IP and port, and does it sound ok)?
Do calls from the problematic mobile into the system sound ok?
If your provider and your PBX support wideband audio, is that a factor (does it work if you disable VoLTE on the mobile)? Is the mobile using Wi-Fi calling? If so, does disabling that make a difference?
If you have a complex network setup (anything other than extensions on the same LAN as the PBX), please describe it.
i echo @Stewart1’s advice here.
Take a look at sngrep or any other capture, and see if you are truly Sending RTP and/or receiving it.
Some platforms only allow symetric RTP - you cant Receive until you send, so port or other issues can result in this symptom
Thanks @Stewart1 and @sholinaty,
It is a very basic setup. Just a local install of FreePBX with 9 extensions.
Even with the mobile phone turned off there is no audio of the voicemail message so it’s not a WiFi calling issue (thanks for the suggestion). Calling from the mobile to the phone system is fine.
I just found out that the problem is only from the Bria extensions outside the office and when it is used as a miscellaneous destination in call flow control. From a handset in the office I can call it fine.
There are three mobiles, all with the same provider and it only happens with one of them.
I compared the RTP destination IP and port and they are correct.
In a past life I worked as an engineer for a cell carrier.
They might be blocking sip/rtp on common ports.
If you are on Bria solo free this might be hard, but Bria solo, teams or enterprise can support sip over TLS and SRTP. That has, in my experience, been able to make it through deep packet inspection / application gateways.
Might not help you, but it might.
Wow, I can’t think of any obvious trouble that matches all your symptoms. Just guessing, perhaps a codec issue, a problem with NATed RTP, a buggy SIP ALG.
Two general approaches:
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Start with the Asterisk log of a failing call, including SIP trace. If that looks ok (especially the codecs and port number in the SDP), we can then analyze a packet capture of the SIP and RTP. If that looks ok, we can capture on your WAN interface and or use Bria debug tools.
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Try to narrow down the possibilities with some simple tests. For example: Route the Call Flow Control to an Announcement that then routes to the Misc Destination. When you call in, that will provide an RTP flow. I’d be very surprised if the caller didn’t hear the announcement, but suspect that he will also hear the mobile. Or, try the Bria on both mobile data, on external Wi-Fi, and on office Wi-Fi.
Please tell us about your system: What country are you in? ISP? Modem make/model? Separate router/firewall, if any? What VoIP-related settings in modem and/or firewall (port forwarding, ALG, etc.)? Is PBX in a virtual machine? If so, which platform and which networking mode? SIP transport for trunk (UDP, TCP, TLS)? Using SRTP? Trunking provider? (If you don’t want to say, let us know what codecs they use, whether they proxy RTP or route directly to the carrier, whether they support wideband audio.) If you have multiple trunks, do calls to the failing mobile number use the same trunk as those to the working mobiles?
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Thanks @Stewart1 and @sholinaty,
@Stewart1, routing it via an announcement was a great idea and it works.
I would love to get to the bottom of this, but I have spent too much time on it already and this is an acceptable workaround.
Thanks a lot for your help, it is very much appreciated.
Kind Regards,
Pieter
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