Incoming SIP calls

[ceejcast]

I have an IP500 V1 with release 6.0.18, and have 2 incoming sip numbers. The first one is working fine but the second one which was recently ported does not reach the pbx. Monitor traces show that the second number has a + inserted at the front of the number string. When I inserted the +, I get the same problem. I tried both settings in the Call Routing Method : Request URI and To Header with no success. Thanks in advance.

 

[Pepp77]

Sounds like the SIP provider is sending it in

DDI: E.164 (canonical with plus)

format.

I have never managed to get this to come in properly either and generally always use

DDI: Canonical without plus

and that just works.

| ACSS SME |

 

[DavidCT]

On your SIP URI, put an asterisk ( * ) in all three fields. Use this URI for incoming calls. This will tell the Avaya to accept all calls on this SIP Trunk.

Then create an ICR based on the DID and destination.

Create a second channel with a unique outgoing ID for making calls. Use whatever your current settings are (Use Credentials... or Use Internal Data). Besure to update your ARS to reflect the second channel's Outgoing group ID.

 

[hairlessupportmonkey]

the IPO will route a +XXXXXXXXXX on the incoming call route.

if you want to do a trunk to trunk transfer send the destination to a #

if you want to match in system shortcodes, prefix the # with a number of your choice, e.g 8#

ignore the error in the validation errors.

then you can match in shortcodes and send out via a trunk of your choice.

stuff i figured out making the IPO a SIP gateway for Lync ;-)

ACSS - SME

 

[ceejcast]

I am using nexVortex as the carrier. The +XXXXXXXXXXX did not work in the incoming call route, but I put it in SIP URI in the Local URI field and put XXXXXXXXXX in the incoming call route. The incoming calls worked perfectly. Putting * in the SIP URI removed the SIP tab in User section which I needed for outgoing cid. Thanks for everyone's help.