[asterisk-users] Help with asterisk and avaya SIP trunking

[Robert Boardman]

Krishna Sumanth Chava wrote:
> Hi * Users,
>  
> I ran into a problem when I was trying to communicate an avaya IP 
> Office talk to asterisk with SIP Trunking. I had successful calls from 
> asterisk to Avaya but not from avaya to asterisk.
>  
> Can someone provide me insight on how to address it or the path to 
> resolve it.
>  
> The error I get is mentioned below: (dialing 32564 from avaya to asterisk)
>  
> "[Nov  6 17:14:23] WARNING[6227]: chan_sip.c:8686 get_destination: 
> Huh?  Not a SIP header (Tel:+32564)?
> [Nov  6 17:14:23] NOTICE[6227]: chan_sip.c:13774 
> handle_request_invite: Call from 'avayanew' to extension 'Tel:+32564' 
> rejected because extension not found."
>  
> A SIP Debug of the packet when this happens on asterisk CLI is
>  
> "<--- SIP read from 10.10.8.2:5060 <http://10.10.8.2:5060> --->
> ACK Tel:+32564 SIP/2.0
> Via: SIP/2.0/UDP 
> 10.10.8.2:5060;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9
> From: avayanew <sip:avayanew at avayanew>;tag=d60c0430c7b26cbd
> To: Tel:+32564;tag=as51355066
> Call-ID: 0182709d8c1d025f42dd3dd767c7e8b7 at 10.10.8.2 
> <mailto:0182709d8c1d025f42dd3dd767c7e8b7 at 10.10.8.2>
> CSeq: 152795667 ACK
> Max-Forwards: 70
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
> Content-Length: 0"
>  
> Note: 10.10.8.2 <http://10.10.8.2> is avaya and 10.10.8.1 
> <http://10.10.8.1> is asterisk
>  
> As I understand, we are getting a Tel URI and a "+" like in e.164 
> format and then the number dialed (32564)from avaya. These errors are 
> coming on asterisk console when I try to dial a call from Avaya IP 
> Phone over its SIP trunk on to the asterisk. We probably have to strip 
> the 'Tel:+', so that the asterisk gets the number and thus follows the 
> dialplan programmed in extensions file.
>  
> Please advise. Any help is appreciated.
>  
> Thanks as always
>  
> Regards
> Krishna
> ------------------------------------------------------------------------
>
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you need to make sure the sip dial command in the ipoffice is set to
dial 9n;
feature dial
code n

in system
the set the dial delay timer to 4 seconds

and the dial delay count to 1

this will allow 4 seconds in between each digit

there is a setting on the ipo to change the TEL:+ setting to url setting

cannot remember wher it is but it in the sip trunk settings


robb