Multiple SIP Lines - Only first one accepts calls 3

[sceaton]

Hello!

IPO500v2 9.1.10.0 build 192

I'm in the US and have a SIP trunk group with Primus. We're on a dynamic IP and they require us to register each DID to accept incoming calls. I had 9 DID's and recently added 30 more. Because each needs to register, I have a Credential completed for each DID. The SIP URI then uses the corresponding Credential to register that DID. An Incoming Call Route then controls where to route calls from that DID.

The problem is, I can define a max of 30 Credentials per line. I need 39. So to get around this, I duplicated the existing SIP line and setup the new block of 30 DID's there. All DID's register correctly but incoming calls to this second SIP Line are returned with 404 Not Found, like this:

 1113749mS SIP Tx: UDP 192.***.*.55:5060 -> 216.181.**.14:5060
                    SIP/2.0 100 Trying
                    Via: SIP/2.0/UDP 216.181.**.14:5060;branch=z9hG4bK1nr7m100bomh2mk1p6k0.1
                    From: <sip:******5210@216.181.**.17;user=phone;Broadsoft=LINGO-81kbujqs025qc>;tag=SD9mu1101-1046964130-1492025583220-
                    Call-ID: SD9mu1101-33b4e5619db7760061cfe206028c82fc-oedv170
                    CSeq: 835965755 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY
                    Supported: timer
                    Server: IP Office 9.1.10.0 build 192
                    To: "*******" <sip:1******9660@primusxtension.com>;tag=f032a114841d2d4d
                    Content-Length: 0
                    
   1113750mS Sip: c0a8030c000006bf 17.1727.1 -1 SIPTrunk Endpoint(f17dc128) Present Call, no match (1******9660) from URI in To header or (1******9660) from request URI
   1113750mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 216.181.**.14:5060 to 216.181.**.14:5060
   1113751mS SIP Tx: UDP 192.168.*.55:5060 -> 216.181.**.14:5060
                    SIP/2.0 404 Not Found
                    Via: SIP/2.0/UDP 216.***.**.14:5060;branch=z9hG4bK1nr7m100bomh2mk1p6k0.1
                    From: <sip:******5210@216.181.31.17;user=phone;Broadsoft=LINGO-81kbujqs025qc>;tag=SD9mu1101-1046964130-1492025583220-
                    Call-ID: SD9mu1101-33b4e5619db7760061cfe206028c82fc-oedv170
                    CSeq: 835965755 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY
                    Supported: timer
                    Server: IP Office 9.1.10.0 build 192
                    Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
                    To: "********" <sip:1*******9660@primusxtension.com>;tag=f032a114841d2d4d
                    Content-Length: 0

If I move one of the new numbers to the original SIP Line (17) It works.
If I move one of the old numbers to the NEW SIP Line (18) it does NOT work.

The original (working) SIP Line is 17. The New SIP line is 18. On a hunch, I changed the new SIP line number from 18 to 16 (so that it's before the old SIP Line) and voila! those numbers WORK. But the OLD SIP line (17) STOPS working.

First, it doesn't make sense that I need to duplicate credentials for every SIP URI, but I can't get IPO to register each number any other way. If the numbers aren't registered, I can't dial in. I've tried wildcard SIP URI's, using the same credentials among multiple SIP URI's, all to no avail.

Second, why can't calls on a subsequent SIP Line (18) find the Incoming Call Route?

As always, thanks for your help!

 

[amriddle01]

That's a ridiculous requirement, you typically register with one set of credentials and then just send a different number if you want to use that as caller ID.
So you either have misunderstood the requirements or they are the worst SIP provider in the world

 

[critchey]

Sadly I ran into this issue with Eartlink provider for SIP. Their own documentation states you must setup each DID as a URI to register. I tried setting it up in a sane way and could not get it to work. Luckily my customer only had 5 DIDs it was not a big issue. My suggestion is to get a new SIP provider.

The truth is just an excuse for lack of imagination.

 

[tlpeter]

You can program multiple uri's on one trunk. You should not create a new trunk for a new uri.

BAZINGA!

I'm not insane, my mother had me tested!

 

[amriddle01]

That's not the issue Peter, its credentials not URIs that forced the second trunk

 

[janni78]

A real SIP trunk provider is a good idea.

They have built their solution on a platform not intended for delivering SIP trunks.
You're registering each DDI as it was a SIP phone in their system.
Seen a couple of these solutions here, but they disappeared from the market quickly.

"Trying is the first step to failure..." - Homer

 

[DirectelTech]

So here is a question you can ask your SIP provider, as I have had the same issue.

Normally on their platform they route the DDI to a specific account, however we managed to get them to set the number to route to a specific SIP IP address that includes the "Live IP" of the site and the SIP registrar. An example of the format I use is as below:

010XXXXXXX@XXX.XXX.XXX.XXX//XXX.XXX.XXX.XX

Format is DDI@Site IP // Call registrar.

I will add that I did experience firewall issues where the firewall did not pass the packets through.

Then you either need a trusted IP on their platform for out dialing or just to register a single account for the trunk service.

Hope this helps

 

[pkdev]

I had a comparable issue.
The issue is that IPO is mixing both line because they use the same IP/port on both end.
If you activate SIP Call RX in monitor, you will see the ID of the line on SIP INVITE. It will always use the line with the lowest ID.

The solution is to use a different port (or IP).
Change the listenning port to 5070 (or whatever you want) and tell your SIP provider to use the new port for the 2nd trunk.

 

[sceaton]

Thanks for the feedback everyone! To close the loop on this:

I circled around with Primus and it turns out they ARE able to allow me to register with the Pilot number of our Trunk Group, which will then send all inbound traffic matching a number on our account to IPOffice (using just that one credential). Because of some technical limitation on how our account was setup long ago, they had to create a "fake" pilot number for me to register to. Our actual BTN was setup as a user and there's a known issue on their platform when changing a user back to a Trunk Group Pilot.

After I registered with the pilot number, calls would arrive, but were being returned with 404 Not found because of "Unallocated (unassigned) number". I had written the ICR every way I could think of but it wouldn't work.

For some reason, (and this took me waaayyy too long to figure out) I needed to change the SIP LINE / SIP ADVANCED / Call Routing option to "To Header" instead of "Request URI". Changing the Call Routing allowed everything to work (and use a wildcard SIP URI).

Thanks again for the help!

 

[mic501]

tlpeter is right.
You should create one SIP trunk with multiple URIs, then you can choose the credentials associated with that URI from the drop down list field called "registration" in the URI tab.