You are sending media to the address and port fl.gg has given, but you are getting nothing from them, at the point where this was captured. That’s before the outbound leg is even attempted. In fact, there is no outbound leg in this log.
OK help! I really thought I had a clue. Here’s whatr I did to collect:
tcpdump -l -v > watch.txt
That command executed as root on my PBX. I really thought that would collect all inbound and outbound packets.
I’m only guessing but could it be I need to add the interface?
tcpdump -l -v -i enp4s0 > watch.txt
I did some research so I hope I’m not jumping-the-gun by posting a new log (using the above command updated with the interface).
watch-landline-to-flowroute-tcpip240323-18-56-00.txt (344.6 KB)
I said no outbound leg, not no outbound traffic.
Your call is an inbound call, which is answered, but fails before Asterisk can send an INVITE to forward the call on elsewhere.
Actually I found your use of inbound and outbound confusing, and was trying to clarify the terms. Your logs show outbound media, but no inbound media, but your text says that inbound media is working, which conflicts with the logs and makes me think you are using inbound in a different way from how I would use it. You have both transmitted and received signalling on the only leg, which is an inbound leg, and the content of the signalling indicates the transmitted signalling is reaching the calling system.
I am definitely feeling confused at this point. Let me walk through it in hopes of better-communicating:
- I dial from my landline to flowroute to myPBX.
- On my landline I get no audio
- The PBX gets no audio - I can’t use dtmf for leave voicemail
- When I place the above call I grab packet by using tcpdump on the server
That’s the only way I know to grab packets.
Additionally:
I perform the same steps on T-Mobile.
- T-Mobile call to flowroute to myPBX
- On my T-Mobile phone I actually hear the correct audio from the PBX
- The PBX gets no audio - I can’t use dtmf or leave voicemail.
- When I place the above call I grab packet by using tcpdump on the server
Does that clear things up any?
Please know I do appreciate your help!
Any assistance you can offer in helping me perform tests, etc. – I’ll do my best.
Thank you very much!
EDIT:
I think I just realized what I am missing. I will post tcpdump of a call initiated on my VOIP phone. I am sorry I didn’t catch that earlier. I will post that ASAP.
New Test:
VOIP phone dialing my landline (POTS)
VOIP->myPBX->flowroute->POTS-to-myLandline
Landline rings but when I answer:
No audio.
watch-VOIPphone-mypbx-to-mylandline-tcpip240323-20-18-12.txt (380.2 KB)
Please find attached the tcpdump of this test.
The logs show audio being sent to flowroute at the address to which they have told you to send it. They show no audio arriving from Flowroute. The fault appears to lie outside of Asterisk.
Flowroute says (lines 374 and 376) send audio, as RTP, using, in decreasing order of preference, µ-law, G.729, and A-law codecs), and RFC 4733 DTMF, to port 28740, on 207.223.67.137.
c=IN IP4 207.223.67.137
m=audio 28740 RTP/AVP 0 18 8 101
You reply (lines 478 and 480) that you want them to send audio to you, as RTP, using µ-law for audio, and RFC 4733 for DTMF, at port number 16304 on AsteriskPubIPaddr
c=IN IP4 AsteriskPubIPaddr
m=audio 16304 RTP/AVP 0 101
You actually send audio to them at the port and address they gave:
20:22:20.868999 IP (tos 0x0, ttl 64, id 4426, offset 0, flags [DF], proto UDP (17), length 200)
p7-1456c.16304 > 207.223.67.137.28740: UDP, length 172
They send no media to you but they have sent an ACK (line 709), confirming that their received the address to which you want media sent:
`ACK sip:AsteriskPrvIPaddr:5060 SIP/2.0`
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You need to look at the SDP and RTP in that one and see if the problem is the same.
First, thank you for creating a clear template I could follow. Using your template here is my review of the latest test log:
/watch-VOIPphone-mypbx-to-mylandline-tcpip240323-20-18-12.txt
228 c=IN IP4 AsteriskPubIPaddr
230 m=audio 15624 RTP/AVP 0 101
400 c=IN IP4 207.223.70.230
402 m=audio 53034 RTP/AVP 0 101
478 20:18:19.141506 IP (tos 0x0, ttl 64, id 2014, offset 0, flags [DF], proto UDP (17), length 200)
479 p7-1456c.15624 > 207.223.70.230.53034: UDP, length 172
Does this mean I should ask Flowroute for help as I’m properly handling the audio and passing it to them?
Ideally you track the packets to the edge of your network first, but otherwise it looks like you need their help.
Update:
I opened a ticket with AT&T
my AT&T landline calling my pbx with no audio.
1st-repsonse “we tested from your landline to the edge of our network and the problem is with the next SP, T-Mobile.”
I opened a ticket with T-Mobile. Wow. They are impossible and clueless and refuse to engage the correct resource–please note I have a business account with T-Mobile (which made absolutely no difference to T-Mobile).
I called Flowroute. Wow–they can be amazing when it comes to support. They confirmed (via pcap) then engaged their peer. The peer performed live test calls/captures with me and verified audio to AT&T’s network. AT&T. Ironic.
Flowroute has actually opened a ticket with AT&T and is persuing this issue, there.
Thank you, again, for all the assisstance here in troubleshooting enabling me to successfully engage SPs.
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