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problem with reinvite

Posted by Had on Mon, 10/19/2009
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Thirdlane platform General Questions

I have Polycom phones on the same LAN, all use the same codec and I want to use reinvites. In sip.conf I enabled canreinvite for all phones. But when I make call between these phones reinvite is not issued and the bridge between these phone is not created. I use 6.0.1.76 MTE and asterisk 1.6.0.13.
Please help if you can.

Thanks

Peter

Submitted by Had on Mon, 10/19/2009 Permalink

Sip debug shows that the call is created with dial options rtT.

information from voip-info.org:

"If the Dial() command contains ''t'', ''T", "h", "H", "w", "W" or "L" (with multiple arguments) Asterisk will not issue a re-invite. "

t: Allow the called user to transfer the call by hitting the blind xfer keys (features.conf)

T: Allow the calling user to transfer the call by hitting the blind xfer keys (features.conf)

Somewhere else I found information that if I use -r option only:

"this means you will not be able to do any transfers from an analog phone connected to an FXS port. However, a SIP phone can "break in" to a call to do a transfer, put the call on hold, etc."

I use sip phones only. Do I need to use tT options in dial command? If not how can I change it? Or is there any other way how to use reinvites?

Thanks

Peter

Submitted by eeman on Mon, 10/19/2009 Permalink

reinvites wont work since most of you calls will first invoke chan_local anyway. You have other features that will require asterisk to stay in the media path such as one touch recording, any call recording, PSTN calls coming in over a SIP trunk and cannot re-invite to an internal (private) IP, hold, call stealing etc...

Submitted by Had on Mon, 10/19/2009 Permalink

I need reinvites only for local calls inside 1 tenant. Outside or incomming calls will still go through asterisk. Calls between tenants are going out from TL to GW so I can setup canreinvite=no for sip trunks.

Hold, transfer... are features which are supported by phones.

I found the option setting in script.include and changed it to r only and reinvites are working. Now it just question what will not work. Transfer and hold works fine but call recording and call stealing will not work. But those are features which not everybody will want to use and If so usually they will use it with outside calls. Is there some other important feature which will not work with reinvites?

Thanks

Peter