NS700 sip Trunk

[b106]

Could someone please clarify something for me
I am trying to set up a sip trunk on NS700
From reading the posts on here I am wondering should I have 2 IP addresses.
I believe my trunk is sip peer trunk
Details I was given have 1 IP address for sip server ,no authentation .permit is the static IP address on my router
Should I have another IP for voice.
Peer or register? is one better than the other
Could someone point me to a link for setting up the NS 700 for sip
Thanks in advance

 

[telephoneguy20]

Contact a Panasonic Dealer, or someone that works with sip trunking.

 

[Fodrod]

No you don't need a second public ip. You just need to port forward the DSP port (16000-16511 by default) So that would be udp 16000-16511 forwarded to 192.168.0.102:16000-16511 (with the default pbx ip). For mrg phones you also need to configure the external static IP in voip properties, but for sip trunks this won't cause issues.

 

[b106]

Thanks Fodrod for clarifying that
I have made progress in that I can make outgoing calls but no incoming- call does'nt ring at all on the pbx
The fact that outgoing calls are ok would suggest the sip side is ok
The sip trunk provider uses 10000 to 20000 for voice whereas the DSP card uses 16000 to 16511
Could this cause problems
Thanking You

 

[Mike_K]

Incoming call is routed via DIL or DDI?

 

[Fodrod]

I don't see how that could possibly cause problems, even if your sip trunk provider absolutely requires those ports, you will notice that the panasonics default ports are entirely with that range anyhow.

 

[obtsystems]

The audio ports do not have to match. The panasonic uses 16000 to 16511 for audio and if the sip provider uses 10000 to 20000 then the will talk between 16000 from the panasonic to 10000 on sip provider. They are only audio ports so if you have two way audio you should be good.

If you are using peer sip trunk with out registration it is important to have nat enabled on the sip trunk card with your fixed IP on it. Again this mostly affects the audio.

Port 5060 must be opened on the firewall to the system IP address and locked to the sip provider ip. Peer trunk requires that the provider can send you options ping to the system and that the system replys for incoming calls. If you do a wireshark trace you should see these coming in.

Best to have the port on the sip card to be changed from 35060 to 5060. To do this you will need to change the sip extension registration port to something like 45060 first. Then restart the system

What firewall is in front of the system. What provider are you using for the trunk. There are settings on the sip trunk card that have to be configured for incoming calls

 

[b106]

Thanks for the replies

Apologies for the delay in replying I am just getting over Covid.

I can make calls from the ns700 and I have audio both ways
I have the lan cable to the NS700 going through a switch with port mirroring on it and I can see the sip messages to and from the NS700
If I try to call my own number (I have 2 channels) I get a 503 message service unavailable
Is there any way to capture the messages coming into the router
The router is a Zyxel router .I dont see any option to set up mirroring on it
My set up is as follows and thanks to all who post here because all my knowledge is gleaned from here

Site

​

UDP port for SIP extension server changed to 5090

Shelf property/Main

​

Sip client port changed to 5060
Nat Traversal set to Fixed IP Address
Nat fixed global ip address set to static ip of the router

Port Properties/Main

​

set ch attribute to basic channel
set port 2 to additional channel for slot 1 ch1
Provider set to Magnet
Sip Server set to ip address of SIP server provided by Magnet
Sip Server port no set to 5060 default
Subscriber No set to full telephone number less the leading zero

Port Property/ Account

​

Username Full telephone number less leading zero
Authentication ID Full telephone number less leading zero

Port Properties/Register

​

Register Ability set to Disable

Port Properties /Option

​

set session timer to 3600

Incoming calls set to DDI

How many numbers are sent by the sip trunk to the PBX --I tried it with 9 digits and also with 4 digits?


On the router SIP server ip to 5060 to 5060 to 192.168.1.101 PBX
SIP Server ip to 10000-10511 tp16000-16511 to 192.168.1.102 DSP

Hope above is of help and your comments and guidance is much appreciated
Thanking You

 

[Mike_K]

Try two options:
- Incoming calls set to DDI=Authentication ID
if not successful, then
- Incoming calls set to DIL to the analog extension.

Perhaps the provider uses the Session Border Controller,
in this case, Nat Traversal must be disable.

 

[b106]

Hello Mike
I have tried both DIL and DDI still no incoming
Hello OBT
You mention some settings in the sip trunk card for incoming calls
Can you clarify those settings please
Thanking You

 

[Mike_K]

Did you turn off NAT Traversal?
Set disable NAT Traversal and look at the message number 503 in the trace has changed?

 

[Tele-tech]

The audio ports do not have to match. The panasonic uses 16000 to 16511 for audio and if the sip provider uses 10000 to 20000 then the will talk between 16000 from the panasonic to 10000 on sip provider. They are only audio ports so if you have two way audio you should be good.

Going from past posts, you seem to know a lot more about Panasonic systems than myself so forgive me if I'm incorrect here, but the ports 16000-16511 are RTP ports. RTP is just the audio. SIP, so the initial connection itself, is usually on port 5060, although common security practice is to change this to another port(5065 is becoming a new norm now, which is counter productive). By the sounds of it, the SIP portion is the issue right now if he cannot initiate incoming calls. The open port range would affect audio if that were an issue, but from what I've experienced in the field that audio range just seems to be for the IP/NT line of Panasonic phones, not the RTP stream related to the SIP channel(if any of that makes sense).

b106 - you said you have port mirroring set up on a switch. When you initiate an incoming call, do you get a message in Wireshark that states the 503 error? Are you able to do a packet trace on your router? If you're seeing it on the port mirror port, that means the call is making it's way through your router and being turned away at your PBX, which points to an issue with the config on the PBX itself.

I haven't had to use NAT for SIP trunking, myself. On your router, make sure SIP-ALG is disabled(it may have a different alias depending on the manufacturer, I haven't worked with Zyxel router before).

Can you post a screenshot of your Slot -> System Property -> Site -> Port Number? Or even just confirm the "UDP Port No. for SIP Extension Server" with us. If this port is different than the port provided by your provider, you will need PAT(not NAT) set up on your router. As long as you don't post your public IP address, there shouldn't be any security issues with that.

Also, can you confirm that your SIP trunk is showing registered on the PBX? Make sure you are using the correct SIP Server Port Number that should have been specified by your provider.

Edit: Forgot to ask, have you contacted your SIP Trunk provider? Who is the provider? They can generally do a trace on a call and see where it's being blocked.

 

[Mike_K]

Tele-tech, b106 can make calls from ns700 and has sound in both ways

 

[b106]

Hello
Attached is a wireshark capture
I have 2 channels on the trunk
Capture is of a call made from line 1 of the NS700 to my own number giving 503 service unavailable message.
I can call out from the NS700 to any other number
If I call in to the ns700 from my mobile there is no sip activity on the lan to the ns700
I have a draytec router on order and I hope to be able to do a capture into the router
The ITSP is not very helpful at the moment
This is a test set up for myself
Thanking you

 

[Mike_K]

The provider may have a problem. Message 503 comes from the provider.