Sip Trunk - Incoming call fail

[RRod]

Dear,
I have installed in a IP Office 500 Release 9.0 with a SIP trunk like with 50 channels. It has to be registered with an username and a password.
The service's supplier asks as a requirement to the outgoing calls that the "From" field of the SIP URI sends the ID's username.

Settings for the outgoing calls:
SIP LINE --> SIP URI --> LOCAL URI: Use Credential Authentication Name
SIP LINE --> SIP URI --> REGISTRATION: 1: Credential Authentication Name

With this settings the outgoing calls are out with no problem.

The problem is in the incoming calls. When an incoming calls enters from the trunk line SIP in the traces, the screen shows the error 404 Not Found. The supplier sends the caller ID (12345678).

Settings for incoming calls:
USER --> SIP --> SIP NAME: 12345678
USER --> SIP --> SIP DISPLAY NAME: PRUEBA
USER --> SIP --> SIP CONTACT: 12345678
ICR --> STANDAR --> LINE GROUP ID: 17
ICR --> STANDAR --> INCOMING NUMBER: 12345678
ICR --> DESTINATION: 5678 (PRUEBA)

Could somebody help me with this problem that I've got?

Thank you very much for your help

 

[amriddle01]

You need to add a URI with a * in all fields, make the outgoing line group unique (you don't want it used) make the incoming group 0 (just easier)
Then you can add your DDIs to Incoming Call Route

 

[critchey]

While amriddle's suggestion is exactly how I want to setup a SIP I have run into at least 2 providers that will not work in this fashion. On one provider for it to work I had to use internal data for both outgoing and incoming calls and route them based on the SIP tab. This is super frustrating because if you want 10 DIDs answered by the main group you would instead have to copy the main group 10 times to have a group for each DID(obviously giving each group a unique extension and name). I have also encountered a SIP provider that I had to create a URI for each DID to get that to work. Each SIP provider is different and some are just a pain to deal with.

 

[derfloh]

Each provider is different is correct...

Check the SIP trunk options. You will find an option like caller ID in From header and play with that. I bet you will be able to handle inbound calls through ICR with the */*/* URI.

 

[critchey]

Well the provider I was referring too was Earthlink. Earthlink has a manual created by Avaya on how to get SIP working with Earthlink as the provider. (newest version posted for is R8.0 I honestly didn't even know Earthlink was still around till I had to deal with this mess...)

https://www.earthlink.com/~/media/aff765f531d9461cb6861addf507e9a0.pdf

First paragraph page 17 about SIP URI

"A SIP URI entry must be created to match each incoming number that Avaya IP Office will
accept on this line."

bit farther down

"Set Local URI, Contact and Display Name to Use Internal Data. This setting allows
calls on this line whose SIP URI matches the number set in the SIP tab of any User"

Page 18 of this document

"Channels 2 and 3 display service numbers, such as a DID number routed directly to voicemail or
DID used for Mobile Call Control. DID numbers that IP Office should admit can be entered into
the Local URI and Contact fields instead of Use Internal Data. The numbers 978-555-1168 and
978-555-1169 will be assigned as service numbers in the Incoming Call Routes in Section 5.10."

So according to a document published by Avaya for Earthlink you must use internal data or the specific DID. I spent hours one day trying to get *s URI working and finally conceeded I could not get it (not saying it can't be done just saying I couldn't figure out how). If you have a way around this absurd way of setting up the system I am all ears lol.

Sorry not trying to hijack this thread. I am just stating that it may be possible, for certain SIP providers, that *s URI may not work. I was also getting a 404 error in monitor. Worse case you can try setting it up in this fashion and see if it starts working.

 

[intrigrant]

As I read it you must create a unique URI for each incoming number within the same IP Trunk.
I am glad we don't have providers like that overhere, what if you have 200 DDI numbers? Insane.

 

[janni78]

You can "only" have 150 URIs so that would be fun.
Seriously, providers that have some home-built implementation of SIP should just be ignored.

But I think it depends what their SIP INVITE looks like, without that we're guessing at best.

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

 

[intrigrant]

Probably one of those SIP Providers running low cost SIP phone only provider software not capable of creating trunks or their billing software cannot handle SIP trunks so they only have billing per number. Sad they still exsist and are allowed to call themself a SIP Provider.

 

[RRod]

Hi, going back to the original issue in question, I tried all the ideas that you proposed, but none of them seemed to be effective.

Could you possibly give me some other advice to solve this problema?

Thank you very much for your help

 

[amriddle01]

You will have done it wrong, post the config

 

[RRod]

Hi, I attached a screm shot of config and a sip trace. With this config, outgoing call does work and incoming call failed.

Thank you very much for your help

[URL unfurl="true"]https://res.cloudinary.com/engineering-com/raw/upload/v1475001735/tips/Sip_Trunk_Config_fpw7xk.docx[/url]

Regards

 

[amriddle01]

You probably don't need STUN, that STUN result is not accurate anyway.

The registrars URL should be where you have the address in the "ITSP domain name" not in the separate registrar field.

The URI is not correct, it would typically be all set to "Use credentials username", you would typically make the "Registration required" 1 or 2 mins to keep the return SIP path open in the firewall.

You need a separate URI one of all *'s in it's own outgoing group but same incoming group.

You don't need the proxy address I doubt, I never have.

The incoming call route usually needs to match how they send numbers, not just the last 4 digits, though sometimes this is OK.

You don't need the "@192.168.200.145" part in the shortcode.

And that's just a quick look at what you've shown , there may be more

 

[amriddle01]

As I don't think you need STUN, then you wouldn't normally use LAN2 as "Network Topology" either, set it to None so it follows IP routes

 

[RRod]

Hi amriddle01, thanks for your help.

I answer (in red) about of the email that you sent:

You probably don't need STUN, that STUN result is not accurate anyway. OK, I’ll change it

The registrars URL should be where you have the address in the "ITSP domain name" not in the separate registrar field. I'm sorry but I don't understand. Could you explain this to me?

The URI is not correct, it would typically be all set to "Use credentials username", you would typically make the "Registration required" 1 or 2 mins to keep the return SIP path open in the firewall. OK, I’ll change it

You need a separate URI one of all *'s in it's own outgoing group but same incoming group. I'm sorry but I don't understand. Could you explain this to me?

You don't need the proxy address I doubt, I never have. OK, I’ll change it

The incoming call route usually needs to match how they send numbers, not just the last 4 digits, though sometimes this is OK. OK

You don't need the "@192.168.200.145" part in the shortcode. OK

And that's just a quick look at what you've shown , there may be more

As I don't think you need STUN, then you wouldn't normally use LAN2 as "Network Topology" either, set it to None so it follows IP routes. OK

Thanks & Regards

 

[RRod]

Hi,

Thank you very much for your help!!

I had to create two main lines SIP (SIP Trunk): one for incoming calls, and another for outgoing calls, so I can adjust the SIP URI according to the provider's requirements.

Greetings!!

 

[AlfonsoSF]

I have write how I have mine.

in credentials I have one credential for the registering. With username, and password and no contact name. Register= true. Timeout = 3 min

Then one credential for each DDI. The same username and password, register= false and in contact ddi@proxyIP

In URI SIP "internal data", in "register" choose the credential for registering (usually 1).
choose incoming and outgoing line numbers .

I wish it's useful for you

 

[intrigrant]

I never have more then 1 URI in a SIP trunk, I just wonder why you guys need more.
Do we have better SIP Providers overhere or what?

 

[janni78]

Some SIP implementations are horrible, but here we'll need one all star URI for incoming and one Use Internal Data due to how IPO handles Diversion Headers.

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

 

[AlfonsoSF]

one URI
so many credentials as DDI numbers

Only the generic credential makes the register.
The DDI credentials are only for receiving calls.

URI needs only the credential for registering.

 

[janni78]

If you don't need to register each number you should only have one URI with all stars and use ICR to route calls.

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