SIP number presentation & SIP installation

[Jimbo2015]

Hi,

We subcontracted a SIP installation to an Avaya engineer as we do not have much experience with SIP installations and it was quite complex as the customer only wanted calls to go out over SIP when they called Germany and they wanted their German number presented when they dialled out.

He has configured this for the customer but it does not appear to be working. Please can someone take a look at the config to check this is correct before we ask their IT guy to check the router settings. Also please can someone check the way he has configured the number presentation for line 17 in the ARS table on the attached config.

The sip lines show as registered on system status.

The engineer was alittle unsure himself of how to present a German number when using a SIP line so I suspect he has done this wrong.

Many Thanks,

 

[joe2938]

I am wondering if your sip provider is even allowing a German number to be out pulsed for your sip line

 

[Jimbo2015]

Yes they have confirmed that we can show any number we like. They do not actually provide us with a number so they have said that we need to display the outbound number

 

[Okkie26]

In the ARS. Line 0049N > telehphone number remove the Z"1105078"S03020924040


Then, Line > SIP Line >SIP URI > on the contact, display name and PAI put 00493020924040

 

[amriddle01]

He's certainly attempted to send the number, not in the way I would do it at all, perhaps your provider supports that method....none of ours do, and that registration setup seems odd

 

[Jimbo2015]

Thanks very much for the help here. Amriddle, please could you elaborate on what is odd about the registration set up so I can point him in the right direction.

 

[Pepp77]

I would probably do it a bit like this

First URI for registration
Second for incoming calls (if ever needed)
Third for outgoing calls to present the German number



| ACSS SME |

 

[Jimbo2015]

Ok thanks Pepp I will try this

 

[Seacats]

Do the trunks work at all?

System on 192.168.99.211/24
IP route 0.0.0.0/0 uses gateway of 192.168.0.1 (should be 192.168.99.x)

 

[intrigrant]

I would not advise to use three URIs on one SIP trunk as it is unnecessery and may cause other errors.
Do as Okkie26 advise, that is the best solution.

 

[Pepp77]

I have about 200 SIP trunks in service that are all programmed the same or very similar way and have no issues at all. I even have systems with multiple SIP trunks all programmed the same way with no issues.

It is the way I was trained to program SIP trunks by an ex Avaya Tier 4 engineer.

| ACSS SME |

 

[Jimbo2015]

Seacats I changed the IP address on the config I sent to be safe, so thats nothing to worry about.

I followed Okkies advice but when they try and call a German number it says call rejected. I have been told this may be due to the ARS table, which was set up as;

Code: 0049N
Feature: dial
Number: 0049N"@sip2.voipyonder.com
Line group ID: 5

It has been suggested that instead of putting N in the code I should put X's for the amount of German digits omitting the first 0 but I have search German numbers and they often have a different amount of digits.

Any advice will be greatly appreciated.

 

[amriddle01]

0049N; Dial . But you may need to increase the dial delay in that ARS table, for that reason alone I would use a unique ARS for the SIP line, I never mix trunk types in ARS as it's silly doing it that way

 

[intrigrant]

@pepp77, the Avaya Tier 4 engineer may be trained by me in the past

There are always multiple ways to program something and all are good if it works.
The reason why I always try to avoid multiple URI's in one trunk is that in the past ( don't know if it has ever changed ) incoming calls on the trunk were rejected with the message "network out of order" if the number of channels on the first URI were occupied or if the number of SIP Channels licenses were al used.
With only one URI that cannot happen.

 

[amriddle01]

I'm fairly certain you can chose what response is sent in that scenario now, however there is no good reason for multiple URIs here, it's just not required and offers nothing more than additional programming for the sake of it

 

[Jimbo2015]

Please can you confirm the reasoning for putting a semi colon after 0049N

Many Thanks,

 

[Jimbo2015]

P.s I created a second ARS table specifically for SIP and to test it I also created a separate short code 9N so that calls with a 9 prefix go out over SIP and calls with ?. Go out over ISDN30 but if they dial out with a 9 prefix it will not even let them type half the full cli before it comes back with a error

 

[intrigrant]

A semicolon tells IP Office to wait for additional digits until the dial timout has expired and then it considers the number as complete and send it to the line, no more additional digits can be dialled if the number is send to the line.
So if you dial a number but wait too long between two digits to complete the number you think you cannot dial the number.

 

[Jimbo2015]

Thanks for the help here. We have now made a successful call. However, we have now configured the system so that calls going out through ?. go out through ISDN30 and calls going out through 9N go out over SIP.

We have made a successful call but most of the time we press 9 to dial out and then half way through dialling the number it says incompatible.

Any ideas why some numbers are working and some not?

Many Thanks,

 

[Okkie26]

Where is the 9 pointing to?

ARS? Then change the dial timeout in that ARS (at least 4 seconds)