SIP Trunking URI Issue 1

[cferrera]

We have set up a SIP Trunk on our 406 unit for testing. Incoming calls work perfectly, the issue is out going. Since the Softswitch is ours we are able to view the invite messages to diagnose the issue.

Basically the invite is being sent from the IP Office to the softswitch with:
invite TEL: +1718XXXXXX

Should be as per the softswitch admin:
invite sip: +1718XXXXXXX

There is an option to turn "use Tel URI" on and off on the SIP Trunk form. As per the help file this will change the invite syntax from sip: to tel:. We have this option off already. we have tried turning it on.

Any quality ideas would be terrific.

Thanks in advance.

Chris Ferrera
ACE

 

[tlpeter]

i did set it up with

ITSP domain name : Provider.net
ITSP ip adres : providers ip adres
Primary authentication name : here i did put the telephne number !!!

I also did have problem with in or outgoing only (not sure if it was in o outgoing)

Setting the primary authentication name to the number it started to work


ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[PlatinumGuy]

I am currently running the V4 tech training course and we had this excat issue on one of the mchines yesterday.

Had to smile and move on cause we couldnt fix it. Turning tel uri on and off made no difference at all. Nor diddeleting and recreating the sip trunk.

ACA - IP Office Implement
ACS - IP Office Implement
ACE - IP Office Implement
ACI - IP Office Implement

 

[cferrera]

Thats embarrassing! Not your fault though. This is either something really easy to fix or a deep coding issue.

 

[groundedagain]

Just had a look at Monitor while making a call and the Invite message definitely says

INVITE sip:0418XXXX@XXXX SIP/2.0

running 4.0.7 on IP500 so unless it is something on the 406 that makes it different.

HTH

Jason Wienert
Brisbane, Australia

http://www.centralcity.com.au

GoldMine, Avaya, ACCPAC CRM

Please remember to thank preople for their valuable input.

 

[PlatinumGuy]

If you look on your sip trunk jason you have a tick box for use tel URI.

Ticking thta changes it from sip: to tel: which is what we are saying, even though the system does not have use tel uri ticked it is trying to send it as a tel uri which wont work.

This feature is for some SIP providers who support this.

ACA - IP Office Implement
ACS - IP Office Implement
ACE - IP Office Implement
ACI - IP Office Implement

 

[cferrera]

We tried it the Tel URI option ticked and not ticked. it still sends out the same way as TEL:+1718XXXXXXX.

I am going to try to kill the sip line and re-create. It seems no one else really has this issue. Maybe it is a fluke. Ill let you know.

Thanks all!

 

[technicaluser4]

Hi,

I had exactly the same problem last week when trying to install SIP for the first time.

Incoming calls work immediatley but no outgoing calls.

I solved the problem by adding the realm in the ARS taking care for outgoing calls through the SIP trunks.
In other words you have to program the following:

Let's Say that the provider's telephone numbers start with 88

Then put in an ARS shortcode like:

88N
Feature: Dial
Telephone 88N"@sipprovider.com"

For me it worked straight away!

I hope this helps.

 

[cferrera]

Thanks technicaluser4. That idea worked. I change my 1N to include "@sipprovider.com" . Actually we used the ip address and that worked.

 

[alaxman]

Hi,

We're seeing the same issue, but the suggested fix doesn't seem to work. I was wondering if you could clarify a couple of points:

-> When you say "88N" or "1N" I assume you are referring to the digits that are prefixed before the outbound number. Is this correct?

-> Were you able to get registrations to work? We don't see the IP Office sending any requests out.

Thanks!

 

[technicaluser4]

Yes, the 88N means that you provide some sort of prefix so that you can be sure that the IPO is sending the call throught the SIP line group id.

As regards to registration logging, make sure that the SIP filter is on in Monitor Application becasue by default it is off.


If you did so then if I were you I would try to connect the IPO directly on the public Internet to make sure that calls are coming in and out and eliminate any uncertainty about NAT/firewalls.

Cheers!

 

[alaxman]

One more question on the setup:
Is the "@sipprovider.com" with or without the enclosing ""?

Regarding registrations:
- we don't see anything on the softswitch side either. This would seem to imply that IP office is not sending registration requests
- is there a way to query the registration status of the SIP line?

Thanks again!

 

[Drew2400]

To answer your first question... Yes, the quotes are necessary.

To insure that I'm registered, I've always watched the monitor as technicaluser4 described above. If you see constant registration, then it's not working. One of the tell-tale signs is to watch for the SIP registration entries and your switch asking for "OPTIONS". You'll see a response from the ISP or switch saying INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, etc. The top of the entry should say OK and RX. That applies to the actual REGISTER command as well.

Hope this helps... Drew

 

[cferrera]

To make it more clear. I am using in this case my SIP Line as my PSTN connection, so when a user dials 9,1-XXX-XXX-XXXX they use a SIP Line.

ARS Entry is:
Code=1N; Telephone number=1N"IPADDRESSofSIPProvider" Feature=Dial, Line Group ID = 1 (same as SIP LINE)

 

[alaxman]

Thanks a lot - we got this working finally.

Two things were holding us up:
-> Needed to setup 'STUN' for registrations (or have the firewall rework SIP messages)
-> We were using ARS directly instead of short codes. Once we switched to short codes, everything worked