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

[Robert Boardman]

Krishna Sumanth Chava wrote:
> HI Robb,
> I had the checked the IP Office and i see that in the SIP Line 
> Settings an option [checkbox] that says (Use Tel URI), which is 
> unchecked. But i still get the Tel:+ in the SIP Header (even when it 
> is turned on or off).
>  
> "you need to make sure the sip dial command in the ipoffice is set to
> dial 9n;
> feature dial
> code n"
>  
> do you mean that i need to program this in the ARS of the avaya IP office?
>  
> i have version 4.1(9) firmware on the Avaya IP small Office. Can you 
> share me on what Firmware version of avaya IP small Office, you got 
> the Asterisk and avaya talking to each other.
>  
> Thanks
> Krishna
>  
>  
>
>  
> On Fri, Nov 7, 2008 at 2:59 PM, Robert Boardman <robb at boardman.me.uk 
> <mailto:robb at boardman.me.uk>> wrote:
>
>     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/>
>     <http://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>
>     > <mailto: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/> <http://10.10.8.2
>     <http://10.10.8.2/>> is avaya and 10.10.8.1 <http://10.10.8.1/>
>     > <http://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
>
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sorry its something like

dial 9n;
feature dial
code n"@192.168.0.1"


where the ip address is the asterisk box

robb