Hi all
I have installed an Asterisk version 13.17.2 with FreePBX 14.0.5.25.
The server is on Hyper-V virtual machine on a provate network and it has a private IP 192.168.0.18
I have configured two SIP phones on two extensions and i can make calls between these two.
I had an issue with my SIP provider, but now Asterisk is able to register to them.
My server is behind a SonicWall. I think i have configured correctly the access rules and NAT policies.
This is what i did:
I allowed port 5060 and 5061 UDP, and made a NAT policy which translates it from my public IP to my SIP server’s private IP. This policy is reflexive.
Same way i allowed range 6000-60000 UDP (which my provider asks for media) and made a NAT policy for it.
In this stage this is what happens:
My SIP server registers to my provider.
If i call my public SIP number, the selected extension rings and display’s the callers CID correctly.
When i pick up the extension, i hear nothing and the caller’s phone keeps ringing.
Same way, if i try to dial out, it cannot connect and after a long silence the Asterisk tells me that “The number is not answering”.
I have both inbound and outbound routes set up, also set up the dialling patterns.
I think that it must be Firewall or NAT related, but everything seems te be fine.
If someone can help me, i can provide any log, debug info, tcpdump, etc.
Thanks in advance,
The PBX tries 3 times to retransmit the SIP/2.0 200 OK to my SIP provider.
Problem is, that the To does not look to me.
I hope it is not a problem if i mention my provider.
So, the To is like this:
To: "Our_company_name"sip:our_phone_number@iristel.net:5060;tag=as28fe625c
this iristel.net domain is different of the one of the SIP server and will not answer.
In my trunk i don’t have this anywhere, both my host and fromdomain settings are different, an IP of the proxy, provided by the SIP provider.
Why does it want to transmit the 200 OK to this host?
It would have been part of the previous SIP messages. That is not where the packet itself is going, you can see where based on the IP address+port it is being sent to. Providing the full trace would allow a better answer from people here.
I can provide a full trace, but i am not sure that it is OK to share all the sensitive info (external IP, phone number, extensions, names, etc). Please advise.
Good morning!
I made a debug log for the event when i call our number, the extension rings and i cannot answer it.
It is quite big so i better attach it is a text file:incoming.txt (28.5 KB)
I replaced the following:
Your 200 OK to the external network (208.x.x.x) is not getting ACKed. I don’t know if the To line would be the issue as they are the ones to have created this line in the first place.
The ACK would be sent to the information in the Contact header. I’m unable to open the file because of the Discourse CDN but if that is not correct, then the ACK would not be received.
The SIP proxy set by me is 208.89.128.80
The 200 OK still wants to go to @iristel.net and NOT to sbc1.iristel.net
I don’t know where from comes that @iristel.net because I did not set it.
Of course in the mean time i am trying to solve the problem with their technical support, but it is still unclear to me if the problem is on their side or on mine.
I think you’re still looking at the text of the To header, which is something that they inserted (it is their name for you), and this really shouldn’t matter in most cases, but also, the fact is that the To is unchanged from the initial request that they generated except for the addition of the tag.