Help with SIP Trunks - Optus Evolve Voice

[jensenpos]

Hi,

Hoping for some help with configuration with a setup I am trying to do.

I am in Australia.

We have had Optus Evolve Voice SIP Services installed. They have devlivered these services via a Optic link and terminated into a CISCO router which they provide. There is a PORT on on the back of the CISCO router to which I am "told" to connect an Ethernet cable to.

I already have 3CX setup and operational for use on internal extention calls. I have tried configuring the SIP Trunks, but I have very limited knowledge of the setup of this.

Has anyone ever done this setup before? Anyone give me any pointers?

Regards

Nathan

 

[lneblett]

They should have provided some credentials and perhaps some basic set-up info. My suggestion is to take those same credentials and modify them so that they are invalid and post what they provided on the forum using the same terminology so that we can see what they gave you and possibly provide some advice without compromising your details.

So, if they gave an "AuthID = 12345", you publish "AuthID = ABC987"and so on to include any IP addresses or domain names. We only care about the field names not the variables that are used to populate same. Also include any instructions they may have provided.

 

[jensenpos]

Thanks.

These are the details they have provided: (real settings are being replaced with X)

OPTUS SIP SIGNALLING IP ADDRESS - 123.X.X.X/32
RTP MEDIA OPTUS ADDRESS RANGE - 211.X.X.X/26
SERVICE NUMBERS XXXXXXXX00 - 99
MRS ROUTER LAN IP - 192.168.0.253/29
SIP SIGNALLING ADDRESS OF CUSTOMER PBX - 192.168.0.249/29
RTP MEDIA GATEWAY IP ADDRESS OF CUSTOMER PBX - 192.168.0.249/29

This is all they have given me.

This is the situation as I see it:

They have given me a fibre connection into the building. This has been terminated into a MRS (this the term they use to describe this CISCO router they have given us). The CISCO router has an IP address on our internal LAN of 192.168.0.253.

I have connected a network cable from the network port on the back of the CISCO router (which they labelled LAN for me) and connected the other end to our network switch.

From here I am really not sure what details i should be entering into the SIP settings in 3CX. Optus tell me I should be entering their SIP signalling address into the PBX, but I can't really see how 3CX at an address of 192.168.0.249 will know to communicate with the OPTUS SIP signalling address via 192.168.0.253. I have an separate internet gateway address already, and the SIP signalling address is only available via this MRS router they refer to. Do I need to get a second NIC installed in the 3CX computer, one for internet connection and the other for connection to the SIP network?

Optus have provisioned 10 SIP trunks through this connection and given us a block of 100 direct dial numbers (hence the number range above)

For clarification, we have two fibre connections now. One 10/10 link for internet, and a 2/2 link for SIP trunks.

Hope someone can make sense of this information.

Nathan

 

[lneblett]

In the SIP Server hostname or IP you should insert the 123.X.X.X/32 address. Then, 5060 in the SIP Server port field.

You may need to install the same numbers from above into the proxy fields, but I would start off by leaving the proxy address blank as it can be populated later if needed. I would go ahead and fill out the proxy port to be 5060 as well, as it really won't matter one way or the other.

Then, take the first service number they provided and insert that into both the external number and the authentication ID fields. Set the system to the number of calls for that trunk and then go to the advanced tab and check the box for PBX provides audio, do not require registration, and use this IP - where the IP is the public IP for the SIP trunks. You can start with this and expand by replicating for the other SIP trunks as needed. What you will need to do is to set the other numbers up as DID and set the inbound routes accordingly. You may want to take the first trunk and then take the next 9 numbers (total of 10) and test to see if the calls come thru. You may then want to take the same trunk and delete one of the 9 added numbers and insert say number 50 or other outside of the first 10 to see if they are associating the call paths sequentially to groups of 10. Make a call to that 50th number and see if the calls goes thru. I am guessing that you may need to set up 10 SIP trunks so use the same IP and ports for each and then use the external numbers that correlate to numbers 11, 21, 31, 41, 51, 61, 71, 81 and 91 with number 1 already having been established. I am thinking that OPTUS is really only concerned with the IP, but 3CX will want to see the other.

As fas as another NIC, I do not know as it may depend on your router and how the gateways are set. Is it a multi-WAN capable router, can it handle aliases, can it route based upon protocol, etc? Did they provide 2 connections to you to plug into a router? If the system is dedicated for 3CX and if the SIP trunk will also handle other traffic, then it may not be a big deal. Of course it will be slower and may impact call quality if you use the system for other things while calls are occurring, but if not and you do not mind doing updates and other tasks beyond call traffic during off-hours or non-peak periods, then it may be fine. Just have to do it one step at a time.

For the moment focus on the SIP and select the CODEC suitable for Australia (g711a or g729, the 711 being preferred as 729 is only supported to 1/2 of your 3CX license as it is still under a patent and 3CX had to pay royalties).

 

[jensenpos]

Thanks for your help.

Using your instructions for the SIP side of things I am now able to make and receive calls. However I have no audio either way on incoming or outgoing calls. At the moment, I am simply using the iOS 3CX app for testing. Although I have tried an IP phone with the same result.

I have had to put a static route on the PC running 3CX to route traffic for 123.X.X.X via 192.168.0.253 (being the local address of their CISCO router.

G.711 a-law is the CODEC they have told me to use.

What is the difference between the SIP signalling address, and the RTP MEDIA address range?

Based on the IP Address information of 123.X.X.X/32 (for SIP signalling) and 211.x.x.0/26 (for the RTP media range) I have issued two static routes as below:

route add -p 123.x.x.x mask 255.255.255.255 192.168.0.253
route add -p 211.x.x.0 mask 255.255.255.192 192.168.0.253

Someone has told me that the RTP media addresses may be why I cannot get audio?

 

[lneblett]

RTP is indeed the audio. Usually this issue is associated to a firewall getting in the way and not allowing the stream to get thru. If you have Windows firewall or other 3rd party firewall in between you either need to disable it or make the needed adjustments to the rules. You can do a search for "firewall ports" on the 3cx site and they will have an article on which ports and protocol are needed. You may want to disable it as a test first. If it works, then you know the issue and can make the changes.

You should not really need to do a route. The way it works is that when you make a call, you will be sending them the details of which port and IP you expect to see the stream on your side. This will be in the Invite and the SDP. They will then tell you where they want to see your stream coming back. The process is reversed when they initiate the call to you (inbound). Assuming that you have the system set to allow the RTP stream to traverse the firewall, then you will get the audio.

If it does not work, then it is something else and a wireshark capture may be needed in order to see how the call is being set-up. I had one client last month who connected to a local independent telephone co-op that had a similar issue. Come to find out after a few days of frustration that they were actually using a SBC for me to connect into at the their site rather than the SIP server and they wanted the VIA information to reflect the public IP as they were not using the C= info.

It could be that Optus is not allowing the RTP stream thru which the capture will show.

 

[NickD_3CX]

To try to split the issue into 2 parts, I would first try calling from the iOS client to *777 and talk into the phone. This is an echo extension and you can test if you have 2 way audio between the iOS client and the PBX. IF you hear your voice back its ok, if not the issue may not in fact be between the provider and the PBX.

If the call to the echo extension is OK, you can then focus on the other part of the call.

 

[jensenpos]

Hi All,

Thanks for the replies and the help. @NickD_3CX - audio locally was always working between extension, the issue was only when I tried an external call. After not being able to work it out easily and having no real experience in PBX and SIP, I called several 3CX dealers locally here. After speaking with several, I found a very helpful and knowledgable person from C2 Communications in Melbourne. Issues all sorted in a couple of hours and everything is working perfect now. I did have to keep the static routes on the 3CX PC.

Also it seems 3CX doesn't like having its LAN IP address changed after it is setup and installed. Some records to do with the 3CX tunnel were still reflecting the old LAN IP address.

Nathan

 

[NickD_3CX]

Glad to hear it has been sorted.

For the record, yes, changing the LAN IP of the server post-installation can cause a lot of issues and we advise avoiding it.

Locking thread as solution was found.