[asterisk-users] Help with asterisk and avaya SIP trunking
Krishna Sumanth Chava
kschava at gmail.com
Sun Nov 9 21:28:48 CST 2008
Hi Guys,
Thanks that did help to resolve my issue. i tried the ."@10.10.8.1" and it
worked and i had a successful call but i have the following 2 concerns.
1. We have voice communication from avaya to asterisk now but avaya
is forcing asterisk to use only codec G723. if i disable G723, it says no
compatible codecs. While the calls from asterisk to avaya are being accepted
as "alaw"
2. I am having issues with DTMF. DTMF is not being recognized or being sent
from avaya to asterisk.
I had connected an Analog phone to the POTS line of the IP Office for this
experiment.
Also i am having hard time for detecting Hangups.
Please advise.
Any help is appreciated as i am new to avaya IP office and am much familiar
with asterisk.
Regards
Krishna
On Sat, Nov 8, 2008 at 12:28 PM, Robert Boardman <robb at boardman.me.uk>wrote:
> 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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> >
> >
> > ------------------------------------------------------------------------
> >
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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
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
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>
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